JP2009180893A - Content playback device controlled with acoustic signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a content playback device controlled with acoustic signals for providing a guide terminal inexpensive and easy to operate using with a general purpose terminal. <P>SOLUTION: When a person holds this device materialized as a portable terminal device and approaches an exhibit, an acoustic signal input means 210 obtains sound emitted from a speaker installed near the exhibit as an acoustic signal, and an ID extraction means 220 extracts an embedded content ID from the acoustic signal. A playback list holding means 230 obtains a content playback list of the content corresponding to the extracted content ID from a content storage means 200 and holds it. Meanwhile, a playback means 240 refers to the playback list holding means 230, extracts the content from the content storage means 200, and plays back it according to the playback list. A different content is played back at each time when the person approaches the exhibit by making the sound in which the ID for specifying the content related to the exhibit is embedded be emitted from the speaker near the exhibit. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention is a guide system for exhibits using portable terminals in museums, art galleries, theme parks, and event venues, and related information linked to the exhibits is displayed using images (still images / moving images), text information, and voice information. Related to the technology to service.

  In recent years, in a museum or the like, a method of lending a voice guide terminal to a visitor has become widespread. This audio guide terminal allows random access to audio files, and a method of listening to explanations at the pace of the visitor by giving a button on the terminal to the number specified for the explanation point in the exhibition hall. ing. However, the conventional voice guidance terminal has a problem that it is difficult for an elderly person and a child to operate buttons, and a dedicated terminal, so that it is heavy and expensive.

To solve the problem of difficult button operation, a system has been developed that uses the ID number of the explanation points, infrared rays, and IC tags to send to the terminal, and automatically switches the explanation on the terminal side when the visitor approaches the work. (See Patent Document 1). Moreover, since it is a dedicated terminal, a proposal has been made to realize voice guidance with a general-purpose mobile phone for the problem of being heavy and expensive (see Patent Document 2).
JP 2005-316851 A Utility model registration No. 3124080

  However, in the technique described in Patent Document 1, it is necessary to install an ID transmission facility in the exhibition hall, and the terminal side must also be provided with an infrared ray or an IC tag reader. . Further, the technique described in Patent Document 2 has a problem that the operability is adversely affected although the apparatus cost is reduced.

  Accordingly, an object of the present invention is to provide a content reproduction apparatus controlled by an acoustic signal, which can provide a guide terminal that is easy to operate and inexpensive using a general-purpose terminal.

  In order to solve the above problem, in the present invention, an acoustic signal in which a content ID for specifying content is embedded is input, the content ID is extracted from the input acoustic signal, and the extracted content ID is supported. A content playback device that searches for content composed of a plurality of frames and reproduces each frame in a predetermined order, the content composed of a plurality of frames including at least one of a still image, a moving image, and audio A content storage means for storing a reproduction list indicating a reproduction order in each content in association with the content ID, an acoustic signal input means for recording a predetermined section of the acoustic signal and creating an audio file, ID extracting means for extracting the content ID embedded from the audio file, and the ID extracting means In accordance with the playlist holding means for holding the playlist corresponding to the content ID extracted by the above, and the playlist stored in the playlist holding means, the content is acquired from the content storage means and the frame in the content is played back. Provided is a content playback apparatus controlled by an acoustic signal having playback means for performing the above-described operation.

  According to the present invention, the sound in which the content ID is embedded is recorded, the content ID is extracted, and the content corresponding to the content ID is reproduced in a predetermined order. It is possible to provide a guide terminal that is easy to operate and inexpensive. In particular, when content is stored in a portable terminal and a sound is embedded from the vicinity of each exhibit with a content ID corresponding to the exhibit, a different content is played each time it moves to the vicinity of each exhibit. And serve as a guide for each exhibit.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Embedding content ID)
In the content reproduction apparatus according to the present invention, a sound generated by reproducing an acoustic signal with a music player is acquired as an acoustic signal, and a content ID embedded in the original acoustic signal is extracted from the acoustic signal. Process. Therefore, first, embedding of the content ID into the acoustic signal will be described. FIG. 1 is a functional block diagram showing a configuration of an embedding device for embedding a content ID in an acoustic signal. In FIG. 1, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a frequency component changing means, 40 is a frequency inverse converting means, 50 is a modified acoustic frame output means, 60 is a storage means, and 61 is an acoustic signal storage section. , 62 is an additional information storage unit, 63 is a modified acoustic signal storage unit, 70 is a bit array creation unit, and 80 is a conversion table creation unit. The apparatus shown in FIG. 1 can deal with both a stereo sound signal and a monaural sound signal. Here, a case where processing is performed on a stereo sound signal will be described.

  The sound frame reading means 10 has a function of reading a predetermined number of samples as one frame from each channel of the original stereo sound signal to be embedded with additional information. The frequency conversion means 20 has a function of generating a spectrum by frequency-converting the frame of the acoustic signal read by the acoustic frame reading means 10 by Fourier transformation or the like. The frequency component changing unit 30 extracts a plurality of spectrum sets corresponding to a predetermined frequency range from the generated spectrum, and based on the bit arrangement created by the bit arrangement creating unit 70 from the additional information extracted from the additional information storage unit 62. The function of changing the state of the spectrum set is provided. The frequency inverse transform means 40 has a function of generating a modified acoustic frame by performing frequency inverse transform on a plurality of spectra including the changed spectrum set. The modified sound frame output means 50 has a function of sequentially outputting the generated modified sound frames.

  The storage means 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded as additional information, an additional information storage unit 62 that is configured as a bit string and stores additional information embedded in the stereo acoustic signal, and additional information. It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after embedding, and stores various information necessary for other processing. The bit array creating unit 70 has a function of extracting additional information from the additional information storage unit 62 and referring to the conversion table creating unit 80 for each word of the additional information to create a corresponding bit array. The conversion table creating means 80 assigns Nw (> Nw) bits of a Hamming code having a Hamming distance of at least 4 to all registration ranks of 2 Nw powers that can be taken by Nw bits. It has a function of creating a code conversion table in which bit registration ranks are associated with Nh-bit Hamming codes. Here, the registration order indicates a value when Nw bits are expressed in decimal. When Nw = 7, the registration order is expressed by 0 to 127.

  The additional information is information that should be added to the acoustic information and embedded, and is specifically a content ID that identifies the content. In this embodiment, Nw and Nh are set to Nw = 7 and Nh = 12, respectively. In the present embodiment, since the ASCII code is adopted as the code format of the additional information, Nw = 7 in the additional information, and 7 bits are one word. The bit array created by the bit array creating means 70 is 12 bits, and after the bit array is created, this is processed as one word. Each component shown in FIG. 1 is actually realized by installing a dedicated program in hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

  Next, the processing operation of the embedding device shown in FIG. 1 will be described. The sound frame reading means 10 reads a predetermined number N of samples as one sound frame from each of the left and right channels of the stereo sound signal stored in the sound signal storage unit 61. The number N of samples of one acoustic frame read by the acoustic frame reading means 10 can be set as appropriate, but is desirably about 4096 samples when the sampling frequency is 44.1 kHz. Therefore, the acoustic frame reading means 10 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames.

  There are A type and B type as acoustic frames. In the A-type acoustic frame and the B-type acoustic frame, the next sample after the last sample of the preceding acoustic frame of the same type is set as the first sample. The A-type and B-type sound frames are set by overlapping a predetermined number (2048 in this embodiment) of samples. For example, if the A type acoustic frame is A1, A2, A3... From the top and the B type acoustic frame is B1, B2, B3... From the top, A1 is samples 1 to 4096, A2 is samples 4097 to 8192, A3. Is samples 8193-12288, B1 is samples 2049-6144, B2 is samples 6145-10240, and B3 is samples 10241-14336. Since the A type and the B type are relative, either one may be first. In other words, A1 is samples 2049 to 6144, A2 is samples 6145 to 10240, A3 is samples 10241 to 14336, B1 is samples 1 to 4096, B2 is samples 4097 to 8192, and B3 is samples 8193 to 12288. May be.

  The frequency converting means 20 performs frequency conversion on the acoustic frame after amplitude conversion, and obtains the spectrum of the acoustic frame. Specifically, frequency conversion is performed using a window function. As frequency conversion, Fourier transform, wavelet transform, and other various known methods can be used. In the present embodiment, a case where Fourier transform is used will be described as an example.

  In general, when Fourier transform is performed on a predetermined signal, it is necessary to divide the signal into predetermined lengths. In this case, if Fourier transform is performed on a signal of a predetermined length as it is, a pseudo-harmonic wave is generated. Ingredients are generated. Therefore, in general, when performing Fourier transform, a signal value is changed using a window function called a Hanning window, and then Fourier transform is performed on the changed value.

  In this embodiment, the window function is used, but the window function to be used is divided into the A type acoustic frame and the B type acoustic frame. In the present embodiment, the first window function W (1, i) and the second window function W (2, i) as shown in FIGS. 2A and 2B are prepared so that the extraction side can easily recognize them. I made it. The first window function W (1, i) is for use with an A type acoustic frame, and has a maximum value of 1 at the position of a predetermined sample number i as shown in FIG. In the rear part, the minimum value is set to 0. Which sample number takes the maximum value depends on the design of the window function W (1, i), but in this embodiment, it is defined by [Equation 1] described later. The Fourier transform for the A type acoustic frame is performed on the product of the window function W (1, i).

  The second window function W (2, i) is for use with a B-type acoustic frame, and has a maximum value at the position of a predetermined sample number i as shown in FIG. 1 is set, and the front portion is set to have a minimum value of 0. Which sample number takes the maximum value depends on the design of the window function W (2, i), but in this embodiment, it is defined by [Expression 2] described later. The Fourier transform for the B type acoustic frame is performed on the product of the window function W (2, i).

  As described above, in the present embodiment, the sound frame is read in duplicate. That is, a predetermined number of samples are redundantly read in the odd-numbered sound frames and the even-numbered sound frames. As described above, the window function used is different between the odd frame and the even frame, but since either the odd frame or the even frame is simply the difference between the odd frame and the even frame, either process may be performed on either. . Therefore, in this specification, one of the odd-numbered frame and the even-numbered frame is referred to as an A-type frame, and the other is referred to as a B-type frame. In the present embodiment, an odd frame is described as an A type frame and an even frame is described as a B type frame. Conversely, an even frame may be an A type frame and an odd frame may be a B type frame.

  In the present embodiment, the window functions W (1, i) and W (2, i) are defined by the following [Equation 1] and [Equation 2]. In FIG. 2, the horizontal axis is the time axis (i). As described later, i is a serial number assigned to N samples in each acoustic frame, and is proportional to time t. 2A and 2B, the vertical axis indicates the amplitude value (level) of the signal. 2A and 2B, the vertical axis indicates the values of the window functions W (1, i) and W (2, i), and the maximum values of W (1, i) and W (2, i). Are all 1.

[Formula 1]
When i ≦ N / 8, W (1, i) = 0.0
When N / 8 <i ≦ 3N / 8, W (1, i) = 0.5−0.5 cos (4π (i−N / 8) / N)
When 3N / 8 <i ≦ 11N / 16, W (1, i) = 1.0
When 11N / 16 <i ≦ 13N / 16, W (1, i) = 0.5 + 0.5 cos (8π (i-11N / 16) / N)
When i> 13N / 16, W (1, i) = 0.0

[Formula 2]
When i ≦ 3N / 16, W (2, i) = 0.0
When 3N / 16 <i ≦ 5N / 16, W (2, i) = 0.5−0.5 cos (8π (i−3N / 16) / N)
When 5N / 16 <i ≦ 5N / 8, W (2, i) = 1.0
When 5N / 8 <i ≦ 7N / 8, W (2, i) = 0.5 + 0.5 cos (4π (i−5N / 8) / N)
When i> 7N / 8, W (1, i) = 0.0

  As is clear from FIG. 2 and [Formula 1] and [Formula 2], the window functions W (1, i) and W (2, i) are asymmetrical to each other. This is for facilitating identification between the two on the extraction side described later.

  In the present invention, since odd frames and even frames are redundantly read by a predetermined number of samples, after embedding information and then restoring to an acoustic signal, the odd frame multiplied by the window function and the window function are multiplied. When overlapping samples of even frames are added, it is necessary to return almost to the original value. For this reason, when the window functions W (1, i) and W (2, i) are added in the overlapping portion of the odd frame and the even frame, it is defined to be a fixed value 1 for all sections.

  When the frequency conversion means 20 performs Fourier transform on the A type sound frame, the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). The window function W (1, i) is used to perform processing according to the following [Equation 3], and real part Al (1, j) and imaginary part Bl (1, j) of the conversion data corresponding to the left channel are performed. ), Real part Ar (1, j) and imaginary part Br (1, j) of the conversion data corresponding to the right channel are obtained.

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

  When the frequency converting means 20 performs Fourier transform on the B type sound frame, the left channel signal Xl (i + N / 2), the right channel signal Xr (i + N / 2) (i = 0,..., N−1) ) Is processed using the window function W (2, i) according to the following [Equation 4], and the real part Al (2, j) and imaginary part Bl of the conversion data corresponding to the left channel are performed. (2, j), real part Ar (2, j) and imaginary part Br (2, j) of the conversion data corresponding to the right channel are obtained.

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

  In the above [Expression 3] and [Expression 4], i is a serial number assigned to N samples in each acoustic frame, and takes an integer value of i = 0, 1, 2,... N−1. Further, j is a serial number assigned in order from the smallest value of the frequency value, and takes an integer value of j = 0, 1, 2,... N / 2-1 like i. When the sampling frequency is 44.1 kHz and N = 4096, if the value of j is different by one, the frequency will be different by 10.8 Hz.

  By executing the processing according to the above [Equation 3] and [Equation 4], a spectrum corresponding to each window function of each acoustic frame is obtained. Subsequently, the frequency component changing unit 30 extracts a spectrum set in a predetermined frequency range from the generated spectrum. In the present embodiment, a range between F1 and F2 is extracted.

  After performing the frequency conversion, the frequency conversion means 20 performs conversion so that the average value of each spectrum set becomes a set value. A so-called amplitude conversion process is performed. The purpose of this amplitude conversion is to reduce the level difference for each frame so that appropriate information can be embedded. Therefore, the set value can be set as appropriate. , 128 is set.

  The frequency conversion means 20 first calculates the conversion magnification when performing the amplitude conversion on the obtained spectrum set. The conversion magnification is calculated by dividing the set value by an average effective value that is an average of effective intensity values of a spectrum set in a predetermined frequency range. Specifically, conversion magnifications Zl (1) and Zl (2) for spectrum sets of the L-channel A-type acoustic frame, the B-type acoustic frame, the R-channel A-type acoustic frame, and the B-type acoustic frame. , Zr (1), Zr (2) are calculated by processing according to the following [Equation 5]. When the target sound frame is close to silence and the square sum of the denominator is less than a predetermined value, the conversion magnification is set to 1.0 and the amplitude conversion is not performed. By performing the amplitude conversion, all the sound frames are in the state of amplitude conversion so that the average effective value of the spectrum set of each sound frame becomes the set value regardless of the signal intensity in the original state. Will be embedded.

[Formula 5]
Zl (1) = Zo / [Σ _{j = m,..., M + 2G−1} {Al (1, j) ² + Bl (1, j) ² }] ^1/2
Zl (2) = Zo / [Σ _{j = m,..., M + 2G−1} {Al (2, j) ² + Bl (2, j) ² }] ^1/2
Zr (1) = Zo / [Σ _{j = m,..., M + 2G−1} {Ar (1, j) ² + Br (1, j) ² }] ^1/2
Zr (2) = Zo / [Σ _{j = m,..., M + 2G−1} {Ar (2, j) ² + Br (2, j) ² }] ^1/2

  In the above [Equation 5], Zo = 2G, and in this embodiment, G = 80, so Zo = 160.

  Further, in the range of j = m,..., M + 2G−1 (corresponding to the frequencies F1,..., F2), Zl (1 for each element of Al (1, j) and Bl (1, j) ), Multiply each element of Al (2, j) and Bl (2, j) by Zl (2), and multiply each element of Ar (1, j) and Br (1, j) The amplitude is converted by multiplying Zr (1) and multiplying each element of Ar (2, j) and Br (2, j) by Zr (2). In the following description, Al (1, j), Bl (1, j), Al (2, j), Bl (2, j), Ar (1, j), Br (1, j), Ar (2, j) and Br (2, j) are values obtained by performing these amplitude conversions.

  The frequency component changing unit 30 performs a process of changing the ratio of the predetermined frequency component for the A type sound frame according to the bit arrangement created by the bit arrangement creating unit 70. In the present invention, the bit array is read bit by bit, and 1-bit information is embedded in a pair of acoustic frames of A type and B type. There are two 1-bit values to be embedded: “0” and “1”. In the present embodiment, these are defined as value 1 and value 2. These can be expressed as code 1 and code 2 in that two types of codes can be embedded. At this time, any one of “0” and “1” may be defined as a value 1 and a value 2 (reference numerals 1 and 2). This is because it is sufficient that one bit embedded on the extraction side can be specified on the extraction side. Therefore, this definition must match between the embedding side and the extraction side.

  Various methods of changing the frequency component are conceivable, but in this embodiment, the principle of sound pulse condensation, which is a human auditory psychological characteristic, is used. Here, a description will be given of the sound pulse concentration. The syllable segregation is an illusion that humans interpolate and hear sounds as if two tracks of high and low are flowing continuously against a pattern in which high and low sounds alternate in time series. It is.

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

  Specifically, when the frequency component of a predetermined frequency range in an acoustic frame composed of a predetermined number of samples extracted from an acoustic signal is changed by the embedding device, the strength and weakness generate a sound pulse fraction. Change to As a result, although it does not sound as if the sound is interrupted by a human, the extraction device can recognize the clear change.

  In the present embodiment, using the principle of sound pulse segregation, the components of the change target frequency band of the acoustic frame are changed to two states, and 1-bit information is embedded. Here, the state of change of the predetermined frequency component of the acoustic frame before and after the embedding process will be described. FIG. 4 shows a state of a predetermined frequency component of each channel 1 sound frame of A type and B type according to the present embodiment. In each acoustic frame shown in FIG. 4, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction.

  In FIG. 4, the frequency region is divided into four in the frequency direction of the vertical axis, but the second and third regions from the top, that is, the frequency band from F1 to F2 is the frequency band to be changed. The uppermost part, that is, the frequency exceeding F2, and the lowermost part, that is, less than F1, are frequency bands not to be changed. That is, in the present embodiment, the intensity of the spectrum set is changed by setting the frequency F1 to F2 as a predetermined frequency range. As shown in FIG. 4 (a), for the change target frequency band of the L channel A type acoustic frame, the spectrum on the high frequency side is represented by L1U, the spectrum on the low frequency side is represented by L1D, and the A type acoustic frame of the R channel is represented. For the frequency band to be changed, the spectrum on the high frequency side is represented by R1U and the spectrum on the low frequency side is represented by R1D. Further, as shown in FIG. 4B, for the change target frequency band of the L-channel B-type acoustic frame, the spectrum on the high-frequency side is represented by L2U, the spectrum on the low-frequency side is represented by L2D, and the B-type of the R-channel is represented. For the frequency band to be changed of the acoustic frame, the spectrum on the high frequency side is represented by R2U, and the spectrum on the low frequency side is represented by R2D.

  In the present embodiment, when the code 1 is embedded, as shown in FIGS. 4C and 4E, the product of the strengths of L1D and L2U and the product of the strengths of R1D and R2U are changed to a relatively strong state. And the product of the intensity of L2D and the product of the intensity of R1U and R2D are changed to a relatively weak state. This state is referred to as “state 1”. When embedding the code 2, as shown in FIGS. 4D and 4F, the product of the intensity of L1U and L2D and the product of the intensity of R1U and R2D are changed to a relatively strong state, and the intensity of L1D and L2U is changed. And the product of the intensity of R1D and R2U is changed to a relatively weak state. This state is referred to as “state 2”. The shaded portions have the same density, indicating that the spectrum is a set for obtaining a product. The darker shaded color indicates a group that is changed to a relatively strong state.

  In the present embodiment, information is embedded by changing the frequency components of the A-type and B-type sound frames into two states as shown in FIGS. 4C, 4E, 4D, and 4F. ing. Since there are two states, this corresponds to an information amount of 1 bit.

  In the present embodiment, the change target frequency bands F1 to F2 are set to “1.7 kHz to 3.4 kHz”. This is due to the following reasons. That is, when a mobile phone having a high degree of spread as voice communication is used as a receiving terminal, the upper limit needs to be 3.4 kHz which is the upper limit of the telephone line band and the mobile phone. This is because the low-pass filter mounted inside the mobile phone corresponds to 3.4 kHz or less in accordance with the telephone exchange. Therefore, the lower limit is set to 1.7 kHz, which is one octave lower than the upper limit of 3.4 kHz. Note that the values “1.7 kHz” and “3.4 kHz” are representative values, and are not necessarily accurate values, and may be slightly deviated from them.

  In the example shown in FIG. 4, the change to the relatively strong state and the weak state has been described. However, the strength can be set according to the situation. As will be described below, the larger the ratio between the two, the higher the accuracy at the time of extraction. However, the ratio of interpolation becomes incomplete, and noise due to discontinuous components is heard during reproduction. On the other hand, as the ratio between the two is equal, the reproduction quality approaches the original sound, but the embedded bits cannot be extracted, and the reproduction quality and extraction accuracy are in a trade-off relationship. For example, when the strong side is set to 100% and the weak side is set to 0%, the sound of the portion to be interpolated due to the sound wave segmentation is 50% of the sound that was played in the original sound signal before the change as shown in FIG. % Has been confirmed.

  Therefore, if the strong side is set to 70% and the weak side is set to 30%, the sound of the portion to be interpolated is almost the same as the sound that was played in the original sound signal before the change as shown in FIG. It has been confirmed that this ratio is the limit that can maintain the extraction accuracy. For this reason, it is preferable to set the intensity ratio of the relatively strong spectrum set and the relatively weak spectrum set as 70% and 30%. In order to realize this, in this embodiment, in a specific process described later, a coefficient α = 0.7 for setting a strong state and a coefficient β = 0.3 for setting a weak state are set. However, when the intensity of the spectrum set to be changed to a strong state is originally small, it is necessary to correct the coefficients α and β. For this reason, the frequency component changing means 30 first executes the processing according to the following [Equation 6] to obtain the intensity ratio γ of the spectrum set to be changed to a strong state with respect to the spectrum set to be changed to a weak state. calculate.

[Formula 6]
E1d = Σ _{j = m,..., M + G−1} {Al (1, j) ² + Bl (1, j) ² }
E2d = Σ _{j = m,..., M + G−1} {Al (2, j) ² + Bl (2, j) ² }
E1u = Σ _{j = m + G,..., M + 2G-1} {Al (1, j) ² + Bl (1, j) ² }
E2u = Σ _{j = m + G,..., M + 2G-1} {Al (2, j) ² + Bl (2, j) ² }
When the embedded data is a value 1, γ = (E1d · E2u) / (E1u · E2d)
When the embedded data is 2, γ = (E1u · E2d) / (E1d · E2u)

  In the above [Expression 6], m is the number of the lower limit component of the change target frequency band, and m + 2G is the number of the upper limit component of the change target frequency band. For example, when 1.7 kHz to 3.4 kHz is set as the change target frequency band, m = 160 and m + 2G = 320. Therefore, the width G of one frequency region is 80.

  Further, according to the value of the intensity ratio γ, the frequency component changing unit 30 corrects the coefficients α and β by executing the processing according to the following [Equation 7], thereby obtaining the coefficients α ′ and β ′. obtain.

[Formula 7]
When 0.01 ≦ γ <1.0, α ′ = α · γ ^−1/2 , β ′ = β · γ ^1/2
When γ <0.01, α ′ = 10.0 · α, β ′ = 0.1 · β

  If γ ≧ 1.0, no correction is performed. Further, when the information to be embedded is “value 1”, the frequency component changing unit 30 executes the processing according to the following [Equation 8] to change the frequency component state to “state 1”, that is, 4 (c) Change to the state shown in (e).

[Formula 8]
j = m~m + G-1 of E for each component (1, j) = {Al (1, j) 2 + Bl (1, j) 2 + Ar (1, j) 2 + Br (1, j) 2} ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
j = m + G~m + 2G- 1 of E for each component (1, j) = {Al (1, j) 2 + Bl (1, j) 2 + Ar (1, j) 2 + Br (1, j) 2} ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = B1 (1, j) · E (1, j) · β / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  When the information to be embedded is “value 2”, the state of the frequency component is changed to “state 2”, that is, FIG. 4D and FIG. Change to the state shown.

[Formula 9]
j = m~m + G-1 of E for each component (1, j) = {Al (1, j) 2 + Bl (1, j) 2 + Ar (1, j) 2 + Br (1, j) 2} ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = B1 (1, j) · E (1, j) · β / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
j = m + G~m + 2G- 1 of E for each component (1, j) = {Al (1, j) 2 + Bl (1, j) 2 + Ar (1, j) 2 + Br (1, j) 2} ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  When the coefficients α ′ and β ′ are obtained by executing the processing according to the above [Expression 7], the coefficients α ′ and β are replaced with the coefficients α and β in the above [Expression 8] and [Expression 9]. 'Is used.

  As described above, the frequency inverse transform means 40 performs the process of obtaining the modified sound frame by performing the frequency inverse transform on the frame spectrum in which the state of the frequency component is changed. Naturally, the inverse frequency conversion needs to correspond to the method executed by the frequency conversion means 20. In the present embodiment, since the frequency transform unit 20 performs the Fourier transform, the frequency inverse transform unit 40 executes the Fourier inverse transform.

  Specifically, for the A type sound frame, the frequency inverse transform means 40 performs the real part Al ′ (1,1) of the left channel of the spectrum obtained by any of the above [Formula 8] and [Formula 9]. j), imaginary part Bl ′ (1, j), etc., real part Ar ′ (1, j), etc. of the right channel, imaginary part Br ′ (1, j), etc. The process according to this is performed and Xl '(i) and Xr' (i) are calculated. For the frequency components that are not modified in the above [Equation 8] and [Equation 9], Al (1, j), which is the original frequency component, is used as Al ′ (1, j). In calculating the inverse frequency transform, for Al ′ (1, j) and Bl ′ (1, j), Zl (1) in [Formula 5] is replaced with Ar ′ (1, j) and Br ′ (1 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (1) in [Formula 5].

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

In the above [Expression 10], 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 10] indicate the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 10], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the A type modified acoustic frame are obtained.

  For the B type acoustic frame, the frequency inverse transform means 40 performs the real part Al ′ (2, j) and the imaginary part of the left channel of the spectrum obtained by any one of the above [Formula 8] and [Formula 9]. Using Bl ′ (2, j), the real part Ar ′ (2, j) of the right channel, and the imaginary part Br ′ (2, j), processing according to the following [Equation 11] is performed, and Xl ′ ( i) and Xr ′ (i) are calculated. Note that the frequency components not modified in the above [Equation 8] and [Equation 9] are represented by Al ′ (2, j), Bl ′ (2, j), Ar ′ (2, The original values Al (2, j), Bl (2, j), Ar (2, j), and Br (2, j) are used as j) and Br ′ (2, j). In calculating the frequency inverse transform, for Al ′ (2, j) and Bl ′ (2, j), Zl (2) in [Equation 5] is replaced with Ar ′ (2, j) and Br ′ (2 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (2) in [Equation 5].

[Formula 11]
Xl' (i + N / 2) = 1 / N · {Σ j Al' (2, j) · cos (2πij / N) / Zl (2) -Σ j Bl' (2, j) · sin (2πij / N ) / Zl (2)} + Xlp (i + N)
Xr' (i + N / 2) = 1 / N · {Σ j Ar' (2, j) · cos (2πij / N) / Zr (2) -Σ j Br' (2, j) · sin (2πij / N ) / Zr (2)} + Xrp (i + N)

  By the above [Equation 11], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified acoustic frame are obtained.

  The modified sound frame output unit 50 sequentially outputs the A type modified sound frame and the B type modified sound frame obtained by the processing of the frequency inverse transform unit 40 to an output file.

  Next, the overall flow of processing of the embedding device shown in FIG. 1 will be described. When the embedding device is activated, the conversion table creating means 80 first creates a code conversion table in which Nw-bit registration ranks are associated with Nh-bit Hamming codes. FIG. 5 shows a flowchart of code conversion table creation by the conversion table creation means 80.

  The conversion table creation means 80 first performs an initialization process (S601). Specifically, the 7-bit code “0” is associated with the 12-bit code “1” and registered in the i (= 0) -th code conversion table, and the initial value of the 7-bit code KF is set to the 1, 12-bit code. The initial value of HF is 2. Subsequently, i = 0 is initially set (S602). Next, a Hamming distance between the 12-bit code HF and another 12-bit code already registered in the i-th code conversion table is calculated (S603).

  If the calculated Hamming distance is less than 4, the value of HF is incremented by 1 (S604), and then the process returns to S602 and the next 12-bit code HF is processed. On the other hand, if the calculated Hamming distance is 4 or more, the value of i is incremented by 1 and updated (S605). If i is less than KF-1, the process returns to S603 to calculate the Hamming distance from the 12th code registered in the code conversion table. If i becomes KF-1 or more after S605, the 12-bit code HF is registered at the position of the 7-bit code KF in the code conversion table, and the values of KF and HF are respectively incremented by one and updated. (S606). If KF is less than 128, the process returns to S602 and the process for the next 7-bit code KF is performed. When KF is 128 or more, since 12-bit codes HF corresponding to all 7-bit codes KF have been registered, the code conversion table creation process is terminated.

  The code conversion table created in this way is shown in FIG. As shown in FIG. 6, in the code conversion table, 12-bit codes are registered in association with each value 0 to 127 that can be taken by the 7-bit code. As can be seen from the binary representation of the 12-bit code, all the 12-bit codes have a hamming distance of 4 or more. In the example of FIG. 6, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  Next, the process of the embedding device after the code conversion table creation process will be described with reference to the flowchart of FIG. Each component which comprises the apparatus shown in FIG. 1 cooperates, and performs the process according to FIG. First, the embedding device calculates a total frame number Cf, which is the total number of sound frames that can be embedded in the sound signal to be embedded (S801). Specifically, after a sound signal is sampled to determine the total number of samples, a process of dividing by the number of samples of one sound frame is performed. Next, additional information compression processing is performed (S802). Specifically, the additional information recorded in decimal notation is extracted from the additional information storage unit 62 and converted into a binary format.

  Here, FIG. 8 shows an input character string of additional information recorded in decimal notation and a compressed character string format after compression. In the present embodiment, the input character string format is ASCII characters. In FIG. 8, “#” in the input character string format indicates a delimiter. “N” is an ASCII character of 0-9. Therefore, the input character string requires the number of bytes corresponding to the number of digits plus a break. “D” in the compressed character string is a 2-digit decimal number from 00 to 99 expressed as a 1-byte binary value from 1 to 100. "C" in the compressed character string is a one-digit decimal number with a separator of # or # 0 to # 9 expressed as a 1-byte binary value of 101 to 110. C is omitted for two decimal digits, and C is a fixed value of 101 for four decimal digits. When such compression is performed, as shown in FIG. 8, in the case of 2 decimal digits to 5 decimal digits, it is understood that 2 bytes to 3 bytes are compressed.

  After the additional information is compressed, the unit character string type number CS is calculated (S803). The unit character string type number CS indicates how many types of unit character strings having one meaning are included in a continuous character string of additional information. The unit character string type number CS is calculated by performing a process of dividing the compressed character string word number BS by the word period BC (CS = BS / BC). Here, the compressed character string word number BS is the number of words of the additional information in the state converted into the binary format in S802. The word period BC indicates the number of bytes after compression corresponding to the unit character string, and in this embodiment, the number of bytes after conversion of all unit character strings is the same.

  Subsequently, the number of repeats Cr per unit character string is calculated (S804). The number of repeats Cr indicates how many times the unit character string is embedded. The repeat number Cr is calculated by performing a process of dividing the total frame number Cf by a value obtained by multiplying the unit character string type number CS, the word cycle BC, and the converted bit number Nh (Cr = Cf / CS · BC · Nh).

  Next, the byte counter, repeat counter, and type counter are initialized (S805). Specifically, all these values are set to “0”. Then, 1 byte located in the byte counter of the character type located in the type counter is read from the compressed bit array (additional information character string) (S806). Then, a process of embedding the read 1 byte in the acoustic signal is performed (S807). Details of the 1 byte embedding process will be described later.

  When the embedding process for 1 byte is completed, the word counter is updated (S808). Specifically, the value of the word counter is increased by 1. Then, it is determined whether the word counter has exceeded the word cycle BC (S809). If the word counter does not exceed the word cycle BC, the process returns to S806 to read the next 1 byte. On the other hand, if the word counter exceeds the word cycle BC, the word counter is initialized and the repeat counter is updated (S810).

  Next, it is determined whether or not the repeat counter has exceeded the repeat number Cr (S811). If the repeat counter does not exceed the repeat number Cr, the process returns to S806 to read the next 1 byte. On the other hand, if the repeat counter exceeds the number of repeats Cr, the repeat counter is initialized and the type counter is updated (S812). Then, it is determined whether the type counter has exceeded the unit character string type number CS (S813). If the type counter does not exceed the unit character string type number CS, the process returns to S806 to read the next 1 byte. On the other hand, if the type counter exceeds the number of unit character string types CS, the embedding process is terminated. 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.

  Here, details of the embedding process for one byte in S807 of FIG. 7 will be described with reference to the flowchart of FIG. The number of bits Nw of one word can be set to an arbitrary number of bits, but as described above, in this embodiment, it is set to substantially 7 bits of the ASCII code.

  In FIG. 9, first, the bit array creation means 70 creates a corresponding bit array by referring to the conversion table creation means 80 for each word of the additional information extracted from the additional information storage unit 62 (S101). Specifically, first, the additional information storage unit 62 is extracted in units of 1 word (7 bits), and a 12-bit bit array composed of corresponding Hamming codes is obtained by referring to the code conversion table shown in FIG. Extract.

  These 12 bits are read into a register in a computer used as an embedding device. Thus, in the additional information storage unit 62, one word is 7 bits, but at the time of embedding processing, processing for one word in the additional information is performed with this 12-bit array.

  Next, the frequency component changing unit 30 performs a process of reading one bit from Nh (= 12) bits held in the register (S102). Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 as one A-type acoustic frame, and the frequency converting means 20 Conversion is performed, and amplitude conversion is performed on the obtained frame spectrum (S103). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 2] is performed using the window function W (1, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (1) and Zr (1) and perform amplitude conversion. Similarly, the acoustic frame reading means 10 reads a predetermined number of samples as one B-type acoustic frame from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61, and the frequency conversion means 20 Conversion is performed, and amplitude conversion is performed on the obtained frame spectrum (S104). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 3] is performed using the window function W (2, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (2) and Zr (2) and perform amplitude conversion. Subsequently, when the frequency component changing unit 30 changes the state of the frequency component of the A type sound frame and the B type sound frame according to the read bit value, the processing according to [Formula 6] and [Formula 7] is executed. Then, the conversion ratios α and β are determined (S105). Using this determined conversion ratio, the frequency component changing unit 30 executes processing according to the above [Equation 8] and [Equation 9] according to the value 1 and value 2 received from the bit array creation unit 70, The state of the component of the frequency band to be changed is changed to either “state 1” or “state 2” (S106).

  Next, the frequency inverse transforming unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the A type acoustic frame is changed by the process of S106. Perform (S107). This inverse amplitude transform is performed by multiplying the spectrum by the inverse of Zl (1) and Zr (1) calculated by [Equation 5]. This inverse frequency transform is naturally performed by the frequency transforming means 20 in S103. It is necessary to support this method. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the right channel such as the real part Al ′ (1, j) of the left channel, the imaginary part Bl ′ (1, j), etc. of the spectrum obtained by any one of the above [Equation 8] and [Equation 9]. Using the real part Ar ′ (1, j), etc., the imaginary part Br ′ (1, j), etc., the processing according to the above [Equation 10] is performed, and Xl ′ (i), Xr ′ (i) calculate. The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file.

  Similarly, the frequency inverse transform unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the B type acoustic frame is changed by the process of S106. Perform (S108). Specifically, the inverse amplitude transform is performed by multiplying the spectrum by the reciprocal of Zl (2) and Zr (2) calculated in [Formula 5], and is obtained by any of the above [Formula 8] and [Formula 9]. Real part Al ′ (2, j), etc. of the left channel of the spectrum, imaginary part Bl ′ (2, j), etc., real part Ar ′ (2, j) of the right channel, etc., imaginary part Br ′ (2, j ) And the like, the process according to [Formula 11] is performed to calculate Xl ′ (i) and Xr ′ (i).

  The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file. When the processing for the two A-type and B-type sound frames is finished for each channel in this way, the frequency component changing means 30 reads the next 1 bit in the bit array (S102). The above processing is executed over all samples of both channels of the acoustic signal. That is, a predetermined number of samples are read as sound frames, and when there are no more sound frames to be read from the sound signal (S103, S104), the process ends. When the processing corresponding to each bit of the bit arrangement (Nh = 12 bits) for one word read in S101 is completed, the process returns from S102 to S101 to read the next word of the additional information and create a bit arrangement. Processing will be performed. When the processing is completed for all the words of the additional information, the processing returns to the first word of the additional information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

  In the present embodiment, the case has been described in which the additional information is converted to a 12-bit bit array using the code conversion table by processing the additional information for one word 7 bits. However, in the present invention, extraction is performed. As long as there is an agreement with the content reproduction apparatus on the side, it is possible to record one word of additional information in units of other number of bits.

  Of the modified acoustic signal obtained as described above, for the portion in which the additional information is embedded, the component of the change target frequency band has only two distributions of the state 1 and the state 2 described above. However, since the components other than the component of the frequency band to be changed remain as the original acoustic signals, various distributions are made based on the setting of the producer. Further, as is apparent from the processing of [Equation 8] and [Equation 9], the components of the frequency band to be changed are changed at the same rate in the left channel and the right channel. Therefore, even at a position equidistant from both speakers, the components in the frequency band to be changed are in a relationship to be amplified without being canceled out, and information can be easily extracted.

  The embedding apparatus has been described above. Here, an embedding and extraction method using amplitude conversion according to the present embodiment will be described with reference to FIGS. 10 to 13. For comparison, FIGS. 10 and 11 show an embedding process and an extracting process when amplitude conversion is not used, and FIGS. 12 and 13 show an embedding process and an extracting process using amplitude conversion in the present embodiment. In these explanatory diagrams, the signal waveform of the acoustic signal assuming that 3-bit information is embedded is shown for the odd and even three pairs of acoustic frames, and the acoustic frames are shown in a state where they do not overlap each other for convenience of explanation. ing. In each figure, the waveform at the left end shows a normal acoustic signal waveform in the time dimension, the horizontal direction is the time axis, the time advances toward the right, and the vertical direction is intensity (amplitude). The waveform at the center or right end represents the sum of spectral components in a predetermined frequency range after frequency conversion as a signal waveform, and is a signal waveform after virtually passing through a bandpass filter. The vertical direction is similarly strong, and is originally significantly lower than the left end, but here, for convenience of explanation, it is enlarged and expressed to the same extent as the left end. Although the levels of the two types of waveforms are originally different, they are represented at the same level here. Actually, there are high frequency components and low frequency components outside the predetermined frequency range as the data after frequency conversion, but these are not shown on the assumption that they are not modified, and are embedded according to FIG. The two signal waveforms are limited to the two frequency band components for performing the above.

  FIG. 10 (a) shows the original sound signal. The first pair of sound frames has a substantially constant intensity throughout, and the center pair of sound frames has a smaller odd frame intensity and an even frame intensity. The last pair of sound frames has a large odd-numbered frame and a large even-numbered frame. FIG. 10B shows the result of frequency conversion performed on such an acoustic signal. As shown in FIG. 4, the embedding target component is divided into upper and lower parts and the component is changed for convenience of FIG. ) Are expressed separately as two waveforms (b-1) and (b-2). The waveform shape after frequency conversion differs for each frequency and is not necessarily similar to that shown in FIG. 10A, but here, it will be assumed that it is the same as before frequency conversion. By embedding 3-bit data [0, 0, 0] based on the method described in FIG. 4 with respect to FIGS. 10 (b-1) and (b-2) (in FIG. 4, bit 0 is a sign). 10 (c-1) and (c-2) are obtained. The first pair can express a pattern as shown in FIG. 4, but the other two pairs of upper and lower ones are required to construct a magnitude relationship in the opposite direction with respect to the original stage. It turns out that an appropriate pattern has not been constructed. FIG. 10 (d) shows the result of frequency inverse transformation performed on the result of such modification. In general, the external shape (referred to as an envelope) of an acoustic signal waveform is determined by low-frequency components having a large energy distribution, which are included in signal components outside a predetermined frequency range in this application, and are modified at the stage of FIG. 10 (c). Therefore, FIG. 10D after frequency inverse transformation has a shape similar to FIG. 10A before frequency transformation.

  FIG. 11 shows how extraction processing is performed on the result of such embedding. 11 (a) and 11 (b) correspond to FIGS. 10 (d) and 10 (c), respectively. In FIG. 11B, when performing bit determination, since the pattern assumed in FIG. 4 is not formed except for the first pair, it is difficult to determine a correct bit.

  Next, a method in which amplitude conversion is introduced in the present embodiment will be described with reference to FIGS. FIG. 12A shows the same original sound signal as FIG. 10A, and FIG. 12B in which frequency conversion is performed in the same manner is the same as FIG. 10B. Here, FIG. 12C shows the result of amplitude conversion performed in units of six frames. In the case of FIG. 12, since the amplitude in each frame is flat in FIG. 12 (a), the whole is flat in FIG. 12 (c). Since the intra-frame variation is followed even in the stage of FIG. 12C, it does not normally become completely flat as shown in FIG. (Actually, since FIG. 12 (c) is frequency dimension data, the frequency dimension data is not flat and fluctuates.) Also, the conversion magnification is set for each frame, and the upper and lower two frequency components are set. Since the conversion is performed at the same magnification, there is usually a significant difference between the upper and lower sides (however, the time axis direction is relatively uniform). On the other hand, when 3-bit data [0, 0, 0] is embedded, FIGS. 12D-1 and 12D-2 are obtained. Since FIG. 12C, which is the original stage, has a flat waveform, it can be seen that an ideal pattern as shown in FIG. 4 can be easily constructed in all frames. Next, FIG. 12E shows the result of inverse amplitude transformation performed by multiplying the inverse of the magnification set for each frame. At this stage, a pattern similar to that shown in FIG. 10C is often generated, but any shape can be used. Finally, when frequency inverse transform is performed, FIG. 12F is obtained, and similarly, the shape is similar to that of FIG. 12A of the original signal waveform.

  FIG. 13 shows how extraction processing is performed on the result of such embedding. FIGS. 13 (a), (b) and (c) correspond to FIGS. 12 (f), (e) and (d), respectively. Although the waveform shape after frequency conversion in FIG. 13B is basically different from that in FIG. 12B, the calculated amplitude conversion magnification is a similar value, and the amplitude conversion is performed at substantially the same magnification. 13 (c) is obtained. If bit determination is performed in the stage of FIG. 13C, an ideal pattern as shown in FIG. 4 is formed in all frames, so that correct bits can be determined.

  In this embodiment, as shown in FIG. 12C, since the signal component to be embedded is converted and embedded so as to be flat in the time axis direction, the high frequency side and the low frequency side are embedded. The probability of an unnatural change that completely reverses the magnitude relationship of the component intensities is reduced, and the extraction accuracy on the extraction side can be increased while maintaining the quality.

(2. Reproduction of acoustic signals)
The content ID, which is additional information, is embedded in an acoustic signal representing music by the embedding device. In the exhibition hall, a sound in which a content ID for specifying content corresponding to the exhibition is embedded is emitted from a speaker installed near each exhibition. There are three forms as shown in FIG. In the example of FIG. 14A, each music player 1 to music player 4 includes one speaker 1 to speaker 4, and the music player 1 to music player 4 are placed near each exhibit 1 to exhibit 4. The installed form is shown. In this example, audio signals in which ID1 to ID4 are embedded are reproduced by music player 1 to music player 4, respectively. At this time, it is preferable that the speakers of each music player be separated to some extent. However, if speakers such as those disclosed in JP-A-2002-204492 are used, the speakers may be brought closer to each other.

  In the example of FIG. 14B, each of the music players 1 to 4 has two speakers, and the music players 1 to 4 are installed near the exhibits 1 to 4 respectively. Is shown. In this example, ID1 to ID8 are embedded in each channel of the two-channel audio signals, and the embedded audio signals are reproduced by music player 1 to music player 4, respectively. In this case, a sound in which different IDs are embedded is emitted from two speakers connected to one music player. Therefore, even when the user approaches the exhibit 1 with the content playback device, different IDs are extracted when the user approaches the speaker 1 and when the user approaches the speaker 2.

  In the example of FIG. 14C, one multi-track music player includes four speakers 1 to 4 and speakers 1 to 4 are installed near the exhibits 1 to 4 respectively. Is shown. In this example, ID1 to ID4 are embedded in each channel of the 4-channel audio signal, and the embedded audio signal is reproduced by a multi-track music player. In this case, when the user holds the content reproduction apparatus and approaches the speaker, the ID embedded in the sound emitted from the speaker can be extracted. In this form, since all the speakers play the same music, it does not get in the way even if the speakers are brought close to each other.

  In FIGS. 14A to 14C, different IDs are embedded for each speaker, but each ID may be further changed with time. For example, in the example of FIG. 14A, when a 4-minute music piece is repeatedly played from each speaker, the speaker 1 sequentially embeds four types of IDs of ID11, ID12, ID13, and ID14 every minute. The speaker 2 sequentially embeds four types of IDs ID21, ID22, ID23, and ID24 every minute. The speaker 3 sequentially embeds four types of IDs ID31, ID32, ID33, and ID34 every minute. Four types of IDs of ID41, ID42, ID43, and ID44 may be sequentially embedded every minute. As a result, four different types of content can be played back even in the same place for a long time.

(3. Content playback device)
Next, the content reproduction apparatus according to the present invention will be described. FIG. 15 is a block diagram showing an embodiment of a content reproduction apparatus according to the present invention. In FIG. 15, reference numeral 200 denotes a content storage unit, 210 denotes an acoustic signal input unit, 220 denotes an ID extraction unit, 230 denotes a reproduction list holding unit, and 240 denotes a reproduction unit.

  The content storage unit 200 stores content to be reproduced, and is realized by a storage device such as a hard disk or a flash memory in the portable terminal. In this embodiment, a combination of a still image, a moving image, and audio is defined as one content, and each content is stored with a content ID. In addition, a reproduction list indicating the reproduction order of frames constituting each content is stored in association with the content ID. In addition, as the content of the content, for example, when used in an exhibition hall, a content related to the exhibit or a description of the exhibit is prepared.

  The acoustic signal input unit 210 acquires and inputs the flowing voice as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. Any microphone can be used, regardless of whether it is monaural omnidirectional or stereo directional, as long as the component in the frequency band to be changed can be detected. Even if it is stereo-directional, only one channel needs to be used. In addition, when information is embedded by the embedding apparatus shown in FIG. 1, the information is reproduced in stereo, but the sound from either the left or right speaker can be input or the sound from both speakers can be mixed and input. Good, there are no restrictions on the location of the microphone. This microphone is not highly accurate, and information can be extracted using a microphone with general accuracy.

  The ID extraction unit 220 extracts the embedded ID from the digital sound signal input by the sound signal input unit 210. Details of the ID extracting unit 220 will be described later. The reproduction list holding unit 230 temporarily holds a reproduction list of content to be reproduced. The reproduction list holding unit 230 is a predetermined storage area secured in the memory. The reproduction unit 240 extracts the corresponding frame of the corresponding content from the content storage unit 200 according to the reproduction list held in the reproduction list holding unit 230 and reproduces it. The reproduction processing itself uses a known reproduction processing technique, and displays and outputs the image (still image / moving image) in the frame from the display means, and outputs the sound in the frame from the speaker.

  Each component shown in FIG. 15 is actually realized by incorporating a dedicated program into a portable terminal such as a PDA or a cellular phone having an information processing function. A portable terminal such as a PDA or a mobile phone has a computer function, and a content reproduction apparatus is realized by executing a built-in dedicated program.

  Next, the processing operation of the content reproduction apparatus according to the present invention will be described using the flowchart of FIG. FIG. 16 shows an example where the content is composed of still images. As shown in FIG. 16, the content playback apparatus of the present invention roughly divides and executes two loop processes, an ID extraction loop and a playback loop. First, the ID extraction loop will be described. The audio signal input unit 210 and the ID extraction unit 220 of the content reproduction apparatus are processed in two systems. By using the two systems alternately, it is possible to capture sound without omission. Audio recorded by each system is recorded in an audio file, and a content ID is extracted therefrom. In FIG. 16, S501 to S503 and S504 to S506 are processes corresponding to the respective systems, and are the same processes.

  When the content playback apparatus is activated, the sound signal input unit 210 records the flowing sound, converts the sound into a digital signal, and starts processing to record the sound file 1 as a digital sound signal (S501). In this recording process, since an arbitrary short section is recorded in real time from the sound that is actually flowing, the same time as that short section is required. In the present embodiment, the recording process is performed in units of 4 seconds. In parallel with this recording process, the ID extraction unit 220 extracts the content ID from the digital acoustic signal recorded in the audio file 2 (S502). Since this ID extraction process can process the acoustic signal for 4 seconds in about 2 seconds, it ends during the recording process of another system. In addition, since nothing is recorded in the audio file 2 only for the first time, the ID extraction process is not performed. Then, the recording process by the acoustic signal input unit 210 started in S501 ends (S503).

  The processes of S504, S505, and S506 correspond to S501, S502, and S503, respectively. In S504 to S506, the acoustic signal input unit 210 and the ID extraction unit 220 perform the same processing on a system different from S501 to S503, respectively. That is, in S504 to S506, while the sound signal input unit 210 performs the recording process for the audio file 2, the ID extraction unit 220 performs the content ID extraction process from the audio file 1.

  The content ID extracted in S502 and S505 is compared with the content ID of the playlist held in the playlist holding unit 230 by the ID extracting unit 220. If the content ID is different, the ID extracting unit 220 extracts the extracted content ID. Is extracted from the content storage unit 200, and the playlist in the playlist holding unit 230 is changed. At this time, the reproduction list holding unit 230 holds the frame ID of the first frame as a reproduction target frame. Specifically, the pointer position of the first frame indicated in the reproduction list is held.

  Next, the playback loop will be described. The processing of S508 to S511 in the playback loop is executed by the playback means 240. First, the reproduction unit 240 refers to the reproduction list holding unit 230, acquires the reproduction target frame ID held by the reproduction list holding unit 230, and extracts the frame corresponding to the frame ID from the content storage unit 200 ( S508). Then, reproduction processing of the extracted frame is performed (S509). After the reproduction process, the reproduction unit 240 updates the reproduction target frame in the reproduction list holding unit 230 to the next frame on the reproduction list (S510). Specifically, the pointer position of the next frame shown in the reproduction list is held. Thereafter, the playback unit 240 stands by until the playback start time of the next frame calculated from the playback time on the playback list held by the playlist holding unit 230 (S511). When the next frame playback start time comes, the process returns to S508, and the playback unit 240 refers to the playback list holding unit 230 and extracts the corresponding frame from the content storage unit 200.

  As described above, recording and ID extraction are repeated in the ID extraction loop. In parallel with this, in the playback loop, the playback list in the playback list holding means 230 is referred to and the corresponding content is played back. Therefore, on the playback loop side, after a predetermined playback unit ends, the content corresponding to the content ID extracted by the ID extraction loop processing is played from the beginning, and a different content ID is extracted by the ID extraction loop processing. In such a case, the content corresponding to the content ID is reproduced from the beginning. As a result, for example, when the content reproduction apparatus of the present invention is used in an exhibition hall or the like and contents by exhibition are stored, every time the user moves around the exhibition hall and goes to the vicinity of a different exhibition, In addition, the explanation can be sent from the top. At this time, the owner of the content reproduction apparatus does not need to perform any operation.

  Next, processing of the acoustic signal input unit 210 and ID extraction unit 220, the state in the playlist holding unit 230, and the timing of parallel processing by the playback unit 240 will be described. FIG. 17 is a diagram illustrating a state of parallel processing by each unit. As shown in FIG. 17, the acoustic signal input unit 210 continuously and repeatedly performs two systems of recording processing (shown as “voice recording 1” and “voice recording 2” in the figure). That is, immediately after the recording process to the audio file 1 by the audio recording 1 is finished, the recording process to the audio file 2 by the audio recording 2 is started, and immediately after the recording process to the audio file 2 by the audio recording 2 is finished. The recording process is performed in an alternating system without interruption, such as starting the recording process to the audio file 1 by the recording 1.

  When receiving an audio file from the acoustic signal input unit 210, the ID extraction unit 220 extracts an ID from the acoustic signal recorded in the audio file. When the acoustic signal input unit 210 is performing recording processing on the audio file 1 in order to perform extraction in two systems in parallel (shown as “ID extraction 1” and “ID extraction 2” in the figure) ID extraction is performed from the file 2, and ID extraction is performed from the audio file 1 when the acoustic signal input unit 210 is performing a recording process on the audio file 2. In the present embodiment, the acoustic signal input unit 210 records an acoustic signal in units of 4 seconds. This recording always takes 4 seconds, but the ID extraction means 220 can process one file in about 2 seconds. Therefore, the acoustic signal input unit 210 continues processing without taking a break from the start of processing, but the ID extraction unit 220 is in a standby state until the next audio file is received from the acoustic signal input unit 210 after extracting the content ID.

  The reproduction list holding unit 230 records a reproduction list corresponding to the content ID extracted by the ID extraction unit 220. If the ID extraction unit 220 fails to extract the content ID, the playlist is not changed. The reproduction unit 240 reproduces the content corresponding to the reproduction list held in the reproduction list holding unit 230. In the example shown in the figure, the reproduction list in the reproduction list holding unit 230 corresponds to the content 2 when the reproduction processing of the frame Fr14 of the content 1 is completed. The reproduction process of the first frame Fr21 is performed.

  Here, an example of the playlist stored in the playlist holding means 230 is shown in FIG. FIG. 18A shows an example of content composed of three still images. The content ID of this content is “Co1”. In FIG. 18A, the frame ID is an ID for specifying a frame. In the playback list, the playback order is defined by the order of the posted frame IDs. Accordingly, the reproduction list in FIG. 18A indicates that the content “Co1” is reproduced in the order of Fr11, Fr12, and Fr13. The image file name is a name of a file in which a still image corresponding to the frame is recorded. The switching method / time indicates the switching method when switching from the previous image and the time required for switching. For example, in the case of Fr11, the entire image is switched from the previous image in an instant. The playback time is the time during which a still image is displayed.

  Specifically, FIG. 18B shows a state of still image reproduction using the information of FIG. First, as shown in FIG. 18 (a), Fr11 is displayed as a cut and the playback time is 6 seconds, so Fr11 is displayed for 6 seconds as shown in the leftmost diagram of FIG. 18 (b). Then, as shown in FIG. 18 (a), Fr11 is switched to Fr12 by wiping, and the required time is 1 second. Therefore, as shown in the second diagram from the left in FIG. Wipe for 1 second. As shown in FIG. 18 (a), the playback time of Fr12 is 8 seconds, so Fr12 is displayed for 8 seconds as shown in the middle diagram of FIG. 18 (b). Then, as shown in FIG. 18 (a), Fr12 to Fr13 are switched by dissolve, and the required time is 1 second. Therefore, as shown in the second diagram from the right in FIG. Gradually changes over 1 second. Then, as shown in FIG. 18A, since the playback time of Fr13 is 5 seconds, Fr13 is displayed for 5 seconds as shown at the right end of FIG. 18B.

  Next, a case where content includes moving images or audio will be described. When the content includes a moving image or sound, unlike the case of a still image, when the reproduction of the moving image or sound is started, the reproduction is executed until the moving image or sound ends. Therefore, since it is meaningless to extract the ID during this period, the recording / ID extraction process is interrupted. FIG. 19 is a flowchart when the content is a moving image or audio. As shown in FIG. 19, the point constituted by two loop processes of an ID extraction loop and a reproduction loop is the same as in the case of a still image. In FIG. 19, the same processes as those in FIG.

  19 is different from FIG. 16 in the processing of S512 to S514 in the reproduction loop. Specifically, after extracting the frame data corresponding to the reproduction list from the content storage means 200 (S508), the ID extraction loop is stopped (S512). This is performed by giving an instruction to the acoustic signal input unit 210 and the ID extraction unit 220 from the reproduction unit 240. Subsequently, the reproduction processing of the moving image / sound that is the extracted content is performed (S513). After the reproduction process, the reproduction unit 240 issues an instruction to the acoustic signal input unit 210 and the ID extraction unit 220 to restart the ID extraction loop (S514). Then, the reproduction target frame ID in the reproduction list holding means 230 is updated (S510).

  Next, the processing of the acoustic signal input unit 210 and the ID extraction unit 220, the state in the playlist holding unit 230, and the timing of parallel processing by the playback unit 240 when the content is a moving image or sound will be described. FIG. 20 is a diagram illustrating a state of parallel processing by each unit. When reproducing a moving image or sound, unlike the case of a still image, when the reproduction of the moving image or sound is started, the process of the recording / ID extraction loop is stopped until the end. Therefore, as shown in FIG. 20, the recording process is stopped while the content that is a moving image / sound is being reproduced, and the sound recording is resumed after the reproduction of the content that is a moving image / sound is finished.

(4. Details of ID extraction means)
Next, details of the ID extracting means 220, which is a component of the content reproduction apparatus of the present invention, will be described. FIG. 21 is a configuration diagram showing details of the ID extracting means 220. In FIG. 21, 110 is a reference frame acquisition unit, 120 is a phase change frame setting unit, 130 is a frequency conversion unit, 140 is a code determination parameter calculation unit, 150 is a code output unit, 160 is an additional information extraction unit, and 170 is an acoustic frame. Holding means 180 is a conversion table creating 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). As the reference frame, A type and B type are set as in the case of embedding. The phase change frame setting means 120 has a function of setting, as a phase change frame, an acoustic frame whose phase has been changed by moving each of the A type and B type reference frames and predetermined samples.

  The frequency conversion unit 130 basically has substantially the same function as the frequency conversion unit 20 shown in FIG. However, since the timing to capture the sound does not start from the beginning of the acoustic signal, in order to identify the correct phase, amplitude conversion and frequency conversion are performed with multiple phases, even if the original acoustic signal is stereo, The difference is that one channel is used.

  When performing the Fourier transform on the A type acoustic frame, the frequency converting unit 130 applies the window function W to the signal X (i−N / 2 + pN / 6) (i = 0,..., N−1). (1, i) is used to perform processing according to the following [Equation 12] to obtain the real part A (1, j, p) and imaginary part B (1, j, p) of the converted data. Have. p is a phase number and takes an integer value of 0 to 5.

[Formula 12]
A (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .cos (2πij / N)
B (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .sin (2πij / N)

  When the Fourier transform is performed on the B type acoustic frame, the frequency converting unit 130 applies the window function W (2) to the signal X (i + p · N / 6) (i = 0,..., N−1). , I) is used to perform processing according to the following [Equation 13] to obtain the real part A (2, j, p) and imaginary part B (2, j, p) of the converted data. ing.

[Formula 13]
A (2, j, p) = Σ _{i = 0,..., N-1} W (2, i) · X (i + p · N / 6) · cos (2πij / N)
B (2, j, p) = Σi _{= 0,..., N-1} W (2, i) .X (i + p.N / 6) .sin (2πij / N)

The frequency conversion unit 130 performs amplitude conversion in the same manner as the frequency conversion unit 20. In performing amplitude conversion, first, conversion magnification is calculated. The conversion magnification is calculated by dividing the set value by the average effective value that is the average of the effective intensity values of the spectrum set in the predetermined frequency range. Although the set value can be determined as appropriate, it is necessary to set the same value as in the case of amplitude conversion at the time of embedding. Therefore, in the present embodiment, the set value Zo needs to be 160. Specifically, the conversion magnifications Z (1, p) and Z (2, p) for the A type acoustic frame and the B type acoustic frame, the A type acoustic frame immediately before these, and the B type acoustic frame. The conversion magnifications Z ₋₁ (1, p) and Z ₋₁ (2, p) are calculated by processing according to the following [Equation 14]. By performing the amplitude conversion, all the sound frames are extracted in a state where the amplitude is converted so that the average effective value of each sound frame becomes a set value regardless of the signal strength in the original state. Will be done. In addition, p is a phase number and takes an integer value of 0-5.

[Formula 14]
Z (1, p) = Zo / [Σ _{j = m,...,} M _{+ 2G−1} {A (1, j, p) ² + B (1, j, p) ² }] ^1/2
Z (2, p) = Zo / [Σ _{j = m,...,} M _{+ 2G-1} {A (2, j, p) ² + B (2, j, p) ² }] ^1/2
Z ₋₁ (1, p) = Zo / [Σ _{j = m,...,} M _{+ 2G−1} {A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ² }] ^{1 / 2}
Z ₋₁ (2, p) = Zo / [Σ _{j = m,...,} M _{+ 2G−1} {A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ² }] ^{1 / 2}

j = m,..., m + 2G-1 (corresponding to frequencies F1,..., F2) and p = 0,. . . , 5, each element of A (1, j, p) and B (1, j, p) is multiplied by Z (1, p) and A (2, j, p) and B ( 2, j, p) is multiplied by Z (2, p), and for each element of A _-1 (1, j, p) and B _-1 (1, j, p) Multiply by Z _-1 (1, p) and multiply each element of A _-1 (2, j, p) and B _-1 (2, j, p) by Z _-1 (2, p) Thus, amplitude conversion is performed. In the following description, A (1, j, p), B (1, j, p), A (2, j, p), B (2, j, p), A ₋₁ (1, j, p) , B ₋₁ (1, j, p), A ₋₁ (2, j, p), and B ₋₁ (2, j, p) are values obtained by performing these amplitude conversions.

  The code determination parameter calculation unit 140 extracts a spectrum set corresponding to a predetermined frequency range from the generated spectrum, calculates an intensity value of each spectrum set, and calculates a code determination parameter using the intensity value. Based on the magnitude relationship of the code determination parameters, a function for determining a predetermined state is provided. As described above, in the present embodiment, since the A type acoustic frame and the B type acoustic frame are set to overlap each other by N / 2 samples, the intensity value and the code determination parameter are calculated for a certain acoustic frame. In this case, it is necessary to consider the reverberation component due to the immediately preceding acoustic frame. However, since the reverberation component is calculated, the accurate one is not always calculated. If the calculated reverberation component is removed, the extraction accuracy may be lowered due to warping. Therefore, in this embodiment, the intensity values E1, E2, E3, E4 and the sign determination parameter C when the reverberation component is not removed, and the intensity values E1 ′, E2 ′, E3 ′, E4 ′ when removed, and A correction code determination parameter C ′ is calculated, and a state that seems to be optimal is determined using these parameters.

Each component corresponding to the B type acoustic frame one frame before the respective components A (1, j, p) and B (1, j, p) of the A type acoustic frame is represented by A ₋₁ (2, j, p), B ₋₁ (2, j, p), one frame corresponding to each component A (2, j, p), B (2, j, p) of the B type acoustic frame The sound frame before the minute is the A-type sound frame before correction. The code determination parameter calculation unit 140 first calculates the basic intensity values E (1, j, p), E (2, j, p), and reverberation components of each spectrum set before calculating the intensity values of each spectrum set. The basic intensity values E ′ (1, j, p) and E ′ (2, j, p) from which the above is removed are calculated by processing according to the following [Equation 15].

[Formula 15]
E (1, j, p) = A (1, j, p) ² + B (1, j, p) ²
E (2, j, p) = A (2, j, p) ² + B (2, j, p) ²
E ₋₁ (1, j, p) = A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ²
E ₋₁ (2, j, p) = A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ²
E ′ (1, j, p) = E (1, j, p) −q · E ₋₁ (2, j, p)
E ′ (2, j, p) = E (2, j, p) −q · E (1, j, p)

  Of the six formulas in [Formula 15], the fifth formula from the top (second from the bottom) is the B type in which the N / 2 samples in the first half overlap when focusing on a certain type A acoustic frame. This is for removing the reverberation component from the acoustic frame. In addition, the sixth expression from the top (first from the bottom) removes the reverberation component from the A-type sound frame in which the N / 2 samples in the first half overlap when focusing on a certain B-type sound frame. Is for. E ′ (1, j, p) ≧ 0, E ′ (2, j, p) ≧ 0, and as a result of processing according to the fifth and sixth equations from the top of [Equation 15], If this happens, set it to 0.

  In the above [Expression 15], q is a coefficient indicating the magnitude of the reverberation component, but this coefficient q has a value less than 1, and as a result of experiment, when N = 4096, q = 0.06, When N = 2048, q = 0.12. When N = 1024, q = 0.24. When N = 512, q = 0.48 is optimal.

  Then, the intensity values E1, E2, E3, E4 of each spectrum set when the reverberation component is not removed, and the intensity values E1 ′, E2 ′, E3 ′, E4 ′ when removed are calculated basic intensity values E (1 , J, p), E (2, j, p), E ′ (1, j, p), and E ′ (2, j, p) are calculated by processing according to the following [Equation 16]. .

[Formula 16]
E1 (p) = Σ _{j = m,..., M + G-1} E (1, j, p)
E2 (p) = Σ _{j = m, ..., m + G-1} E (2, j, p)
E3 (p) = Σ _{j = m + G,..., M + 2G-1} E (1, j, p)
E4 (p) = Σ _{j = m + G,..., M + 2G-1} E (2, j, p)
E1 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (1, j, p)
E2 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (2, j, p)
E3 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (1, j, p)
E4 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (2, j, p)

  For E1 to E4 and E1 ′ to E4 ′, a process of updating using the corresponding components in the past is performed. Specifically, it is calculated by processing according to the following [Equation 17] using Te (1, p) to Te (8, p) recorded in the sum total value registration table for registering past components.

[Formula 17]
E1 (p) = 0.67E1 (p) + 0.33Te (1, p)
E2 (p) = 0.67E2 (p) + 0.33Te (2, p)
E3 (p) = 0.67E3 (p) + 0.33Te (3, p)
E4 (p) = 0.67E4 (p) + 0.33Te (4, p)
E1 ′ (p) = 0.67E1 ′ (p) + 0.33Te (5, p)
E2 ′ (p) = 0.67E2 ′ (p) + 0.33Te (6, p)
E3 ′ (p) = 0.67E3 ′ (p) + 0.33Te (7, p)
E4 ′ (p) = 0.67E4 ′ (p) + 0.33Te (8, p)

  Further, Te (1, p) to Te (8, p) are updated by processing according to the following [Equation 18].

[Formula 18]
Te (1, p) = E1 (p) Te (2, p) = E2 (p)
Te (3, p) = E3 (p) Te (4, p) = E4 (p)
Te (5, p) = E1 ′ (p) Te (6, p) = E2 ′ (p)
Te (7, p) = E3 ′ (p) Te (8, p) = E4 ′ (p)

  Eventually, the intensity values E1, E2, E3, and E4 of each spectrum set considering the past are calculated by [Formula 15], [Formula 16], [Formula 17], and [Formula 18], and correspond to each spectrum set. Intensity values E1 ′, E2 ′, E3 ′, and E4 ′ are calculated by subtracting a value obtained by multiplying the intensity of the spectrum set in the immediately preceding acoustic frame regardless of the type by q.

  Also, the code determination parameter calculation means 140 executes processing according to the following [Equation 19] using the intensity values E1, E2, E3, E4 calculated without removing the reverberation component, and the code determination parameter C Is calculated.

[Formula 19]
1) When E1 (p)> E2 (p) and E4 (p)> E3 (p)
C = {E1 (p) .E4 (p) -E2 (p) .E3 (p)} / {E1 (p) .E4 (p) + E2 (p) .E3 (p)}
2) When E2 (p)> E1 (p) and E3 (p)> E4 (p) B = 1 and provisional determination
C = {E2 (p) .E3 (p) -E1 (p) .E4 (p)} / {E1 (p) .E4 (p) + E2 (p) .E3 (p)}
3) In the case of E1 (p) · E4 (p)> E2 (p) · E3 (p)
C = {E1 (p) .E4 (p) -E2 (p) .E3 (p)} / {E1 (p) .E4 (p) + E2 (p) .E3 (p)}
4) In cases other than the above 1) to 3) B = 1 and provisional determination,
C = {E2 (p) .E3 (p) -E1 (p) .E4 (p)} / {E1 (p) .E4 (p) + E2 (p) .E3 (p)}

  Further, the code determination parameter calculation unit 140 executes processing according to the following [Equation 20] using the intensity values E1 ′, E2 ′, E3 ′, and E4 ′ calculated by removing the reverberation component, and performs correction. A code determination parameter C ′ is calculated.

[Formula 20]
1) In the case of E1 ′ (p)> E2 ′ (p) and E4 ′ (p)> E3 ′ (p) B ′ = 0 and provisional determination,
C ′ = {E1 ′ (p) · E4 ′ (p) −E2 ′ (p) · E3 ′ (p)} / {E1 ′ (p) · E4 ′ (p) + E2 ′ (p) · E3 ′ ( p)}
2) In the case of E2 ′ (p)> E1 ′ (p) and E3 ′ (p)> E4 ′ (p), B ′ = 1 is temporarily determined,
C ′ = {E2 ′ (p) · E3 ′ (p) −E1 ′ (p) · E4 ′ (p)} / {E1 ′ (p) · E4 ′ (p) + E2 ′ (p) · E3 ′ ( p)}
3) In the case of E1 ′ (p) · E4 ′ (p)> E2 ′ (p) · E3 ′ (p) B ′ = 0 and provisional determination,
C ′ = {E1 ′ (p) · E4 ′ (p) −E2 ′ (p) · E3 ′ (p)} / {E1 ′ (p) · E4 ′ (p) + E2 ′ (p) · E3 ′ ( p)}
4) In cases other than the above 1) to 3) B ′ = 1 and provisional determination,
C ′ = {E2 ′ (p) · E3 ′ (p) −E1 ′ (p) · E4 ′ (p)} / {E1 ′ (p) · E4 ′ (p) + E2 ′ (p) · E3 ′ ( p)}

  The code output means 150 determines what is determined to be the optimum phase from the acoustic frames (reference frame and phase change frame) corresponding to one reference frame, and selects a code corresponding to the state of the acoustic frame. It has a function to output. The code determination parameter calculation unit 140 and the code output unit 150 constitute an encoding unit. The additional information extraction unit 160 extracts the binary array output by the code output unit 150 in units of Nh bits, refers to the code inverse conversion table, converts it to the registration rank of Nw bits, and further, a predetermined rule It has a function to extract as additional information having meaning after conversion by the above. The acoustic frame holding means 170 is a buffer memory capable of holding two consecutive reference frames (a total of four reference frames for each channel) for each of the A type and B type for each channel. Similar to the conversion table creation unit 80 shown in FIG. 1, the conversion table creation unit 180 has a Nh (having a Hamming distance of at least 4 or more with respect to all the registration orders of 2 Nw powers that Nw bits can take. > Nw) It has a function of creating a code conversion table in which a registration order of Nw bits and a Nh-bit Hamming code are associated by assigning a Hamming code of Nw bits.

(4.2. Processing operation of ID extracting means)
Next, the processing operation of the ID extracting unit 220 shown in FIG. 15 will be described. When the content playback apparatus is activated, first, the conversion table creating means 180 creates a code conversion table. The code conversion table is created by the conversion table creating unit 180 in the same manner as the conversion table creating unit 80, and the code conversion table as shown in FIG. 6 is obtained.

  Next, the processing operation of the ID extraction unit 220 after the code conversion table creation process will be described. The ID extraction means 220 can be set not to perform error correction when an error is detected by a check code, or can be set to perform 1-bit error correction. Hereafter, the processing operation of the extraction apparatus in a setting in which error correction is not performed will be described with reference to the flowchart of FIG. First, the ID extraction unit 220 initializes the phase determination table S (p), the phase determination log, the phase determination flag, the total value accumulation table Te (1, p) to Te (8, p), and the bit counter (S200). ). The phase determination table S (p) is a table for determining the phase, and p takes an integer value of 0 to 5. The initial value is set to S (p) = 0. The phase determination log records the determined phase, that is, the phase number p for each set of one reference frame and five phase change frames, and 0 is set in the initial state. The phase determination flag is a flag indicating whether or not the phase is fixed, and is set to Off in the initial state. The total value accumulation tables Te (1, p) to Te (8, p) include six types of total values E1 to E4 for the past (byte period × Nh frame) and total values E1 ′ to E4 ′ after reverberation correction. This is stored for each phase, and is set to Te (1, p) to Te (8, p) = 0 in the initial state. For the bit counter, 0 is set as an initial value.

  In this way, when the code conversion table is created and the initial value is set, when the user approaches the speaker in the exhibition hall, the acoustic signal input means 210 records and digitizes the flowing music. Input as a digital acoustic signal. More specifically, the audio input from the omnidirectional microphone (or one channel of the directional microphone) is digitized by the A / D converter.

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 210 as a reference frame (S201). Specifically, reference frames for A type and B type are extracted and read into the acoustic frame holding means 170. The number of samples of one acoustic frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the acoustic frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples for each of the A type and B type as reference frames. The acoustic frame holding unit 170 can store two reference frames of A type and B type for each channel, that is, 2.5N samples. When a new reference frame is read, the old reference frame is discarded. It is supposed to be. Therefore, the sound frame holding means 170 always stores four reference frames (continuous 10240 samples).

  The acoustic frame processed by the extraction device can be divided into a reference frame that is set adjacently without interruption from the beginning, and a phase change frame in which the phase is changed. As for the reference frame, when the A type sound frame and the B type sound frame overlap each other by 2048 samples, after setting sample number 1 to sample number 4096 as the first reference frame, the next reference frame is Sample number 2049 to sample number 6144, the next reference frame is set without interruption, such as sample number 4097 to sample number 8192, and the next reference frame is set to sample number 6145 to sample number 10240. Then, for each reference frame, five phase change frames moved by 1/6 frame (about 683 samples) are set. For example, for the first reference frame, five phase change frames configured by 4096 samples starting from sample numbers 683, 1366, 2049, 2732, and 3413 are set. Subsequently, the frequency conversion unit 130 and the code determination parameter calculation unit 140 specify the phase of each read sound frame, determine embedded information, and output a corresponding code (S202). The format of the information to be output has two formats of value 1 and value 2 corresponding to the case where the additional information is embedded.

  Here, details of phase determination and code determination in step S202 will be described with reference to the flowchart of FIG. First, it is confirmed whether the phase determination flag is On or Off (S301). When the phase determination flag is On, the phase determination process (S303 to S309) is not performed, and only the code determination process is performed (S302). However, since the phase is not fixed in the initial state and the phase determination flag is Off, the candidate code table B (p) is initialized (S303). Candidate code table B (p) records one reference frame and five phase change frames, a phase number of p = 0 to 5 and a binary code obtained from the states of these six acoustic frames. To do.

  Subsequently, the code determination parameter calculation unit 140 performs a code determination process (S302). Here, the details of the code determination process are shown in FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain each window spectrum (S401). Specifically, the processing according to the above [Equation 12] and [Equation 13] is executed, and the real part A (1, j, p), the imaginary part B (1, j, p), and the real part A of the converted data are executed. (2, j, p) and imaginary part B (2, j, p) are obtained.

  Regarding the processing of S401, in practice, after frequency conversion is performed on the A type sound frame, the frequency conversion is performed on the B type sound frame after being shifted by N / 2 samples. For these conversion data A (2, j, p), B (2, j, p), A (2, j, p), B (2, j, p), the above [Formula 14] is followed. The process is executed to perform amplitude conversion (S402).

  By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, after the code determination parameter calculation unit 140 calculates the code determination parameters C and C ′ as described above, the code determination parameters C and C ′ are used to determine the state of the component of the change target frequency band. In other words, a process of determining what value is embedded as a 1-bit value is performed (S403). Specifically, the processing according to the above [Formula 15] to [Formula 19] is executed to calculate the code determination parameters C and C ′. Then, comparing these two, if C ≧ C ′, the candidate code B is set in the candidate code table B (p), and if C <C ′, the candidate code B ′ is set in the candidate code table B (p). Output to.

  As a specific processing procedure of S403, first, the basic intensity values E (1, j, p), E (2, j, p) are utilized by using the first and second expressions of [Formula 15]. ), E1 (p), E2 (p), E3 (p), E4 (p) are calculated using the first to fourth formulas of [Formula 16]. 19], the candidate code B and the code determination parameter C are calculated. Subsequently, reverberation correction processing is performed using the third to sixth expressions of [Expression 15] to obtain basic intensity values E ′ (1, j, p), E ′ (2, j, p). Is calculated. Then, E1 ′ (p), E2 ′ (p), E3 ′ (p), E4 ′ (p) are calculated using the fifth to eighth expressions of the above [Expression 16], and the above [Expression 16]. 20], the candidate code B ′ and the correction code determination parameter C ′ are calculated.

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

[Formula 21]
When C ≧ C ′, S (p) ← S (p) + C
When C <C ′, S (p) ← S (p) + C ′

  Here, returning to the flowchart of FIG. 23, the code determination parameter calculation unit 140 stores the code temporarily determined in the code determination process (S302) in the phase p in the candidate code table B (p) (S304).

  Subsequently, it is determined whether or not the processing corresponding to all the phase numbers p has been completed (S305). This determines whether all phase change frames have been processed for a certain reference frame. In this embodiment, since p takes a value from 0 to 5, if the processing is not performed six times, an acoustic frame having a different phase is set by shifting a predetermined number of samples from the acoustic frame being processed. Return and repeat the process. The case where p = 0 is a reference frame, and the case where p = 1 to 5 is a phase change frame. When the processing corresponding to all the phase numbers p is completed, it is determined that the phase corresponding to the phase number pmax having the maximum value in the phase determination table S (p) is the optimum phase, and the candidate code table B (p The code B (pmax) recorded in () is output (S306).

  Subsequently, the phase determination log is updated (S307). The phase determination log records the determined phase, that is, the phase number p for each set of one reference frame and five phase change frames. Then, referring to the phase determination log, it is determined whether or not the phase has been the same a predetermined number of times in the past (S308). In the present embodiment, this number is 10 times. If the phase has been the same a predetermined number of times in the past, the phase determination flag is set to On (S309). As a result, when the same phase continues for a predetermined number of times, the optimum phase is likely to be pmax. Therefore, the phase determination process (S303 to S309) is not performed, and the code determination process is performed only for the phase number p = pmax. (S302) is performed.

  Returning to the flowchart of FIG. When the phase is determined and the code is output, 1 bit corresponding to the output code value is stored in the buffer (S203). Next, the bit counter is incremented by “1” (S204).

  Next, it is determined whether the bit counter is 11 or less or 12 or more (S205). If the bit counter is 11 or less, the process returns to S201 to perform processing for extracting the next A type and B type reference frames.

  If the bit counter is 12 or more, the 12-bit bit array stored in the buffer is decoded (S206). Details of the decoding process will be described with reference to the flowchart of FIG.

  The additional information extraction unit 160 first performs initialization processing (S701). Specifically, a 7-bit code KF and a minimum hamming distance HD are defined, and a process for initializing KF = 0 and HD = 2 is performed. Subsequently, i = 0 is initially set (S702). Next, a 12-bit bit array stored in the buffer is set to a 12-bit code HF, and a Hamming distance hd with a 12-bit code corresponding to the i-th code conversion table is calculated. If hd <HD, KF = i , HD is updated to hd (S703). Then, the value of i is incremented by 1 (S704), and if the value of i is less than 128, the process of S703 is repeated. The additional information extracting unit 160 calculates the hamming distance hd for all 128 hamming codes recorded in the code conversion table shown in FIG. A reference code KF is obtained.

  When the minimum hamming distance HD is obtained, the process returns to FIG. 22 to determine whether the obtained minimum hamming distance HD is 0 or 1 or more (S207). If it is determined in S207 that the minimum Hamming distance is 1 or more, the process returns to S201 to perform processing for extracting the next reference frame.

  If it is determined in S207 that the minimum Hamming distance is 0, the additional information extraction unit 160 adds 1 bit to the 7-bit reference code KF obtained by the processing of FIG. 25 and outputs it (S208). If the minimum Hamming distance is 0 in S207, there is a high possibility that the first bit in the buffer is the first bit of the word in the additional information. Therefore, by adding bit “0” to the 7 bits from the head, it is output as one word in the ASCII code. When the Hamming distance is 1 or more, there is a high possibility that the 12 bits used for collation are shifted from the word in the additional information. In this case, the first 1 bit is discarded, and processing for obtaining a new 1 bit is performed by the processing from S201 to S204.

  If the minimum hamming distance is determined to be 0, there is a high possibility that the portion is a word delimiter, so if it is really a delimiter, then, if it is extracted 12 bits at a time, all are accurately extracted in word units. It can be performed. On the other hand, even if the minimum hamming distance is determined to be 0, it is a coincidence and may not be a word break. In such a case, there is a high possibility of nonconformity at the next inspection, and the correct division can be accurately determined after repeating several times. In S208, when 1 bit is added to the 7-bit reference code KF and output, the bit counter is initialized to 0 (S209). Then, the process returns to S201 to perform processing for extracting the next reference frame.

  By executing the process shown in FIG. 22 for each reference frame, additional information is extracted. If it is determined in S201 that all reference frames have been extracted, the process ends.

  In the process of S208, the additional information extraction unit 160 first converts the value output by the code output unit 150 according to a predetermined rule and extracts it as meaningful additional information. As the predetermined rule, various rules can be adopted as long as the information intended by the person who embeds the information can be recognized by the recipient, but in this embodiment, the ASCII code is adopted. is doing. That is, the additional information extraction unit 160 recognizes the bit value array obtained from the code determined by the code determination parameter calculation unit 140 and output from the code output unit 150 in units of 1 byte (8 bits), and recognizes this as ASCII. The content ID that is character information is recognized according to the code.

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

  Next, the processing operation of the extraction device in the setting for correcting 1-bit error will be described with reference to the flowchart of FIG. 26 includes a portion that performs the same processing as in FIG. Therefore, parts that perform the same processing as in FIG. Also in the example of FIG. 26, first, initialization processing is performed (S200). In this initialization process, as in the example of FIG. 22, the phase determination table S (p), the phase determination log, the phase determination flag, and the bit counter are initialized, and further, the automatic correction mode is set to OFF.

  Subsequently, as in FIG. 22, after extracting an acoustic frame composed of a predetermined number of samples as a reference frame (S201), the phase of each of the read acoustic frames is specified, and the embedded information is determined. The corresponding code is output (S202). Note that the processing of S202 is as shown in FIGS.

  When the phase is determined and the code is output, 1 bit corresponding to the output code value is stored in the buffer (S203), and the bit counter is incremented by "1" (S204). Then, it is determined whether the bit counter is 11 or less or 12 or more (S205). If the bit counter is 11 or less, the process returns to S201 to perform processing for extracting the next reference frame.

  If the bit counter is 12 or more, the 12-bit bit array stored in the buffer is decoded (S206). Subsequently, it is determined whether the obtained Hamming distance is 0 or 1 or more (S207). If it is determined in S207 that the hamming distance is 0, immediately before the bit string output processing in S208, processing for setting the automatic correction mode to ON is performed (S210).

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

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

  When the automatic correction mode is ON, the additional information extraction unit 160 adds 1 bit to the 7-bit reference code KF obtained by the processing of FIG. 25 and outputs it (S208). Then, the bit counter is initialized to 0 (S209).

  As shown in the example of FIG. 26, in the case of setting for 1-bit error correction, even if a 1-bit error occurs, 1 word of additional information can be output.

(4.3. About phase correction processing)
As described above, when an ID is extracted by the content reproduction apparatus, it is not always possible to read an acoustic signal corresponding to the acoustic frame embedded at the time of embedding. Therefore, the phase of the acoustic frame is shifted and read in a plurality of ways (six in this embodiment), the optimum phase is determined, and a code corresponding to the acoustic frame specified by the phase is output. Yes. For example, in the case of reading in six ways, the top acoustic frame is originally a sample of sample numbers 1 to 4096, but six pieces composed of 4096 samples starting from sample numbers 1, 683, 1366, 2049, 2732, and 3413 Are processed, and a code corresponding to the optimum acoustic frame is output. As described with reference to the flowchart of FIG. 23, in the present embodiment, when the same phase continues a predetermined number of times, the process is performed after that phase is determined.

(5. Number of bits in bit array Nh, when Hamming distance is changed)
In the above embodiment, when embedding / extracting the content ID, when creating a bit array, the bit array to be created is 12 bits (Nh bits) and the hamming distance is 4. However, the number of bits of the bit array Nh The hamming distance can be changed as appropriate. In general, if the bit number Nh of the bit array is increased, the coding efficiency is deteriorated, but the number of error bits that can be corrected is increased. Therefore, when using an extraction device with a relatively high microphone sensitivity and arithmetic processing capability, a bit array with a small number of bits and a hamming distance is created, and an extraction device with a relatively low microphone sensitivity and arithmetic processing capability is used. When used, a bit array having a large number of bits and a hamming distance is created.

  Here, an example in which the number of bits Nh in the bit array is 16 and the Hamming distance is 6 will be described. Even when the number of bits of the bit array is 16 and the Hamming distance is 6, the code conversion table creation by the conversion table creation means 80 is basically performed according to the flowchart shown in FIG. However, HF is a 16-bit code, and after calculating the Hamming distance between the 16-bit code HF and another 16-bit code already registered in the code conversion table in S603, whether to proceed to S604 or S605 The threshold is determined by whether the Hamming distance is less than 6 or more than 6 and is different from the case of the number of bits Nh = 12, and the Hamming distance 4.

  The code conversion table created in this way is shown in FIG. As shown in FIG. 27, in the code conversion table, 16-bit codes are registered in association with values 0 to 127 that can be taken by 7-bit codes. As can be seen from the binary representation of the 16-bit code, all the 16-bit codes have a Hamming distance of 6 or more. In the example of FIG. 27, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 16-bit code are shown. Code bit strings are registered in association with each other.

  After the code conversion table creation process, the embedding process is performed according to the flowchart of FIG. At this time, processing for 16 bits is performed as one word.

  At the time of extraction, the conversion table creation means 180 first creates a code conversion table. After the code conversion table is created, additional information is extracted by the processing according to FIGS. When handling a 16-bit code, in S205 in FIGS. 22 and 26, a determination is made based on whether the bit counter is 15 or less or 16 or more. In S701 of FIG. 25, initialization is performed with the minimum hamming distance HD = 3, and in S703, HF is defined as a 16-bit code. Further, the process proceeds to S212 in FIG. 26 when the minimum hamming distance is 3 or more.

(6. Method of embedding information more reliably when the signal component is small)
As described so far, in the embedding and extraction of the content ID in the present embodiment, it is possible to embed information even if the original signal component has a portion close to silence. Of course, this is sufficient, but in the present invention, it is possible to add a process for embedding information more reliably. Specifically, before performing the frequency conversion, weak white noise that is difficult to hear is superimposed on the acoustic signal.

  In this case, the acoustic frame reading means 10 shown in FIG. 1 has a function of adding or subtracting the same value in units of a predetermined number of samples within a predetermined amplitude range after reading a predetermined number of samples as one acoustic frame. Yes. In this embodiment, with respect to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1), an amplitude range of −32 to +32 (Xl (i), Xr ( i) is a value between −32768 to +32767), and a uniform random number H is generated, and the same value H is used between K (K = 5 in the present embodiment) samples. The processing as shown in FIG. 5 is executed to update the values of Xl (i) and Xr (i).

[Formula 22]
Xl (i) ← Xl (i) + H
Xr (i) ← Xr (i) -H

  That is, white noise is generated in one acoustic frame by executing the processing according to the above [Equation 22] over N samples. The white noise generation process is performed immediately after the acoustic frame extraction process of S103 and S104 in the flowchart of FIG.

  As described above, even when white noise is generated, the configuration of the ID extraction means 220 on the extraction side is the same as that in FIG. 21, and the processing operation is according to the flowcharts in FIGS. Are the same.

(7. When creating a sign reverse conversion table during extraction)
In the above embodiment, the ID extraction unit 220 performs the process shown in FIG. 25 with Nh (12 or 16) bits stored in the buffer so as to obtain the corresponding 7-bit reference code KF. However, in this case, since the processing shown in FIG. 25 is executed in units of one word, the processing load increases. In order to reduce such a processing load, it is desirable to create a code reverse conversion table for converting the obtained Nh bits into Nw bits. Here, a case where such a code reverse conversion table is used will be described.

  In this case, the conversion table creation unit 180 in the ID extraction unit 220 has a function of creating a sign reverse conversion table in addition to the function of the conversion table creation unit 80, and the additional information extraction unit 160 follows the function shown in FIG. Instead of the function of performing the processing, a function of referring to the code reverse conversion table is provided. Specifically, the additional information extraction unit 160 refers to the code reverse conversion table instead of executing the processing according to FIG. 25 in S206 of FIGS.

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

  When the content playback apparatus is activated, first, the conversion table creating means 180 creates a code reverse conversion table. The code reverse conversion table is created by the conversion table creating unit 180 by first performing the same processing as the conversion table creating unit 80 to create a code conversion table and then using this code conversion table.

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

  By executing the process shown in FIG. 25 for all of HF = 0 to 4095, a code reverse conversion table as shown in FIG. 28 is created. As shown in FIG. 28, in the code inverse conversion table, for each value 0 to 4095 that can be taken by the 12-bit code HF, the Hamming distance is registered in association with each other, and the Hamming distance becomes 1 or less. Corresponding 7-bit code KF is registered. In the example of FIG. 28, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  When Nh = 16, the process shown in FIG. 25 is executed for all of HF = 0 to 65535, whereby a code reverse conversion table as shown in FIG. 29 is created.

  Then, when performing the processing of S206 in FIG. 22, if Nh = 12, the corresponding 12-bit code KF and the Hamming distance are obtained by referring to FIG. 28 with the obtained 12 bits, and Nh If = 16, the corresponding 16-bit code KF and the Hamming distance are obtained with reference to FIG. 29 with the obtained 16 bits.

(8. Others)
The preferred embodiments of the present invention have been described above. However, the present invention is not limited to the above embodiments, and various modifications can be made. For example, in the above embodiment, the number of samples of one acoustic frame is N = 4096, but N = 2048, 1024, 512, etc. may be set. As a result, the number of sound frames per same time is doubled, quadrupled, and quadrupled, and 2 to 8 times of information can be embedded as a whole.

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

  Further, the embedding of the content ID by the embedding device and the extraction of the content ID by the ID extracting unit 220 are not limited to the above-described embodiment, and JP-A-2006-195061, JP-A-2006-323246, JP-A-2008-. Known techniques such as those disclosed in Japanese Patent No. 15416 can be used.

It is a functional block diagram of an embedding device. It is a figure which shows the time direction window function used by this invention. It is explanatory drawing of the principle of the sound pulse fractionation which is a human auditory psychological characteristic. It is a figure which shows the state of a change of the component of the change object frequency band in one Embodiment of this invention. It is a flowchart which shows the process outline | summary of code conversion table preparation. It is a figure which shows an example of the code conversion table of 7 bit code and 12 bit code. It is a figure which shows the mode of the intensity | strength change of each acoustic frame at the time of embedding in this invention. It is a figure which shows the energy distribution with respect to a frequency, and the embedding area | region of the past and this invention. It is a flowchart which shows the detail of a process of S807 of FIG. It is a figure for demonstrating the embedding process which does not use amplitude conversion. It is a figure for demonstrating the extraction process which does not use amplitude conversion. It is a figure for demonstrating the embedding process using amplitude conversion. It is a figure for demonstrating the extraction process using amplitude conversion. It is a figure which shows the form which emits the sound which embedded content ID from the speaker installed in the vicinity of each exhibit. It is a block diagram which shows one Embodiment of the content reproduction apparatus which concerns on this invention. It is a flowchart which shows the processing operation of a content reproduction apparatus. It is a figure which shows the mode of the parallel processing by each means. 6 is a diagram showing an example of a playlist stored in a playlist holding unit 230. FIG. It is a flowchart which shows the processing operation of the content reproduction apparatus in case a content is a moving image or an audio | voice. It is a figure which shows the mode of the parallel processing by each means in case a content is a moving image or an audio | voice. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a flowchart which shows the process outline | summary in the setting which does not perform error correction of the apparatus shown in FIG. It is a flowchart which shows the detail of the phase determination and code | symbol output of S202 of FIG. It is a flowchart which shows the detail of the code | symbol determination process of S302 of FIG. It is a flowchart which shows the process for obtaining the corresponding 7-bit code KF and the minimum Hamming distance HD when the 12-bit code HF is specified. It is a flowchart which shows the process outline | summary in the setting which performs the error correction of 1 bit of the apparatus shown in FIG. It is a figure which shows an example of the code conversion table of a 7 bit code and a 16 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 12 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 16 bit code.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Acoustic frame reading means 20 ... Frequency conversion means 30 ... Frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified acoustic frame output means 60 ... Storage means 61 ... Acoustic signal storage unit 62 ... additional information storage unit 63 ... modified acoustic signal storage unit 70 ... bit array creation means 80 ... conversion table creation means 110 ... reference frame acquisition means 120 ... phase Change frame setting means 130 ... frequency conversion means 140 ... code determination parameter calculation means 150 ... code output means 160 ... additional information extraction means 170 ... acoustic frame holding means 180 ... conversion table creation Means 200... Content storage means 210... Acoustic signal input means 220... ID extraction means 230. Stage 240: Reproducing means

Claims (8)

Input an audio signal in which a content ID for specifying content is embedded in advance, extract the content ID from the input audio signal, and search for content composed of a plurality of frames corresponding to the extracted content ID And a content playback device for playing back each frame in a predetermined order,
Content storage means for storing a content composed of a plurality of frames including at least one of a still image, a moving image, and audio, and a reproduction list indicating a reproduction order in each content in association with the content ID;
An audio signal input means for recording a predetermined section of the audio signal and creating an audio file;
ID extraction means for extracting the embedded content ID from the audio file;
Playlist holding means for holding a playlist corresponding to the content ID extracted by the ID extracting means;
Reproduction means for obtaining content from the content storage means and reproducing frames in the content according to a reproduction list held in the reproduction list holding means;
A content playback apparatus controlled by an acoustic signal. In claim 1,
If the content ID extracted from the audio file is different from the content ID of the playlist held by the playlist holding unit, the ID extracting unit extracts the playlist held by the playlist holding unit. A content reproduction apparatus controlled by an acoustic signal, wherein the content ID is updated with the reproduction list of the content ID and the frame ID of the reproduction target frame held by the reproduction list holding means is returned to the first frame. In claim 1 or claim 2,
The sound signal input means includes a first sound signal input means for creating a first sound file and a second sound signal input means for creating a second sound file, which are alternately executed in time series. Configured,
The ID extracting means is a first ID extracting means for extracting a content ID from the first audio file, and is a second ID for extracting a content ID from the second audio file, which are alternately executed in time series. ID extraction means,
The first ID extraction means is executed in parallel with the second acoustic signal input means, and the second ID extraction means is executed in parallel with the first acoustic signal input means. A content playback apparatus controlled by an acoustic signal. In any one of Claims 1-3,
In the case where the content is composed of a plurality of still images, the reproduction unit is executed in parallel with the acoustic signal input unit and the ID extraction unit asynchronously, and is controlled by an acoustic signal Playback device. In any one of Claims 1-4,
When the content is composed of a plurality of still images, the playlist includes information including a switching method from the immediately preceding still image, a switching time, and a still image playback time for each frame, A content playback apparatus controlled by an acoustic signal, wherein the playback means controls the playback timing of each frame according to information. In claim 5,
As a switching method from the immediately preceding still image, a dissolve method is used in which the immediately preceding still image, the still image to be reproduced and the luminance value are mixed at a predetermined ratio, and the mixing ratio is changed while changing over time. A content playback apparatus controlled by an acoustic signal. In any one of Claims 1-3,
When the content includes a moving image or sound, the playback means stops execution of the acoustic signal input means and the ID extraction means during playback, and the playback means executes alone. A content playback apparatus controlled by an acoustic signal.   A program for causing a portable terminal device to function as the content reproduction device according to any one of claims 1 to 7.

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