JP2008170828A - Extracting device for information from sound signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an extracting device for information from a sound signal that can perform high-speed extraction processing, even for a general portable terminal, and the like, having low voice recording performance and processing capability. <P>SOLUTION: An embedding device performs embedding processing for information to a source sound signal (a) sampled at 44.1 kHz to obtain a sound signal (b), having the information embedded. After acquiring an analog sound signal, the extracting device samples the sound signal at 22.05 or 8 kHz to digitize it (c). One sound frame is set for the sound signal to have the same time with a sound frame at the embedding (d), and the extraction processing is performed. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

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

  Conventionally, as a service to provide music attribute information that allows you to know the title of the music that is playing, you can query the broadcast station for the date and area of the broadcasted music, Services that record and match melodies registered in the database have been put into practical use. Recently, a method of embedding music attribute information such as a song title and artist information in an acoustic signal as an inaudible digital watermark has been proposed (see, for example, Patent Documents 1 to 4).

  In the methods described in Patent Documents 1 to 4, the amount of information that can be embedded is small, the sound quality is deteriorated to some extent, watermark information is lost by various signal processing, and watermark detection is difficult for analog copies. There is a problem. Therefore, the present applicant has proposed a method of embedding attribute information (additional information) by changing the ratio of the low frequency component of the acoustic signal according to the bit value of the attribute information (see Patent Documents 5 and 6).

  In the inventions described in Patent Documents 5 and 6, both acoustic signals sampled at a frequency of 44.1 kHz and a quantization bit number of 16 bits are set to 4096 samples as one acoustic frame, and 1-bit information is stored for each acoustic frame. Although the embedding process is performed, sampling is performed at the sampling frequency of 44.1 kHz and the quantization bit number of 16 bits at the time of extraction as in the embedding process. Audio recording performance of 1 kHz or higher and equivalent signal processing capability (for example, a CPU having a clock number of 1 GHz or higher) are required. There is no problem when the extraction device is realized by a general PC, but when the extraction device is realized by a portable terminal or the like, it is necessary to select a model having a voice recording performance and a CPU processing capability equivalent to those of a PC. (For example, there are a lot of small PCs and the like realized by a casing of the same level as a portable terminal, so it is not impossible to realize them by selecting them.)

In addition, the present applicant prepared a separate high-performance server for processing in real time using a portable terminal or the like, and obtained the sample by sampling with a portable terminal at a sampling frequency of 44.1 kHz and a quantization bit number of 16 bits. A method has also been proposed in which a sound signal is transferred to a server, information is extracted by the server, and the extracted information is transferred to a portable terminal (see Patent Document 7).
JP-A-11-145840 JP-A-11-219172 Japanese Patent No. 3321767 JP 2003-99077 A JP 2006-235359 A JP 2006-323246 A Japanese Patent Application No. 2005-31755

  However, the reason why mobile terminals that are widely used have relatively low needs for performing high-definition music recording or performing computationally intensive processing, and because of cost reduction, weight reduction, and battery consumption saving, Voice recording performance and CPU performance (especially floating point arithmetic and main memory capacity) are kept low. Therefore, when the extraction device described in Patent Documents 5 and 6 is realized by a general portable terminal, there is a problem that extraction processing cannot be performed in real time. In addition, in the invention described in Patent Document 7, since a process of reciprocating the network is added in addition to the original extraction process, even if the extraction process itself can be executed in real time on the server side, the mobile terminal side in the current network environment It is difficult to obtain a real-time response.

  Therefore, an object of the present invention is to provide an apparatus for extracting information from an acoustic signal that can perform extraction processing at high speed even in a general portable terminal or the like having low voice recording performance and processing capability.

  In order to solve the above-described problem, in the present invention, additional information is embedded for each acoustic frame composed of a predetermined number N of samples in an inaudible state with respect to an acoustic signal digitized at a predetermined sampling frequency Fs. A device for extracting additional information from an analog sound signal reproduced in this manner, wherein a predetermined section of the analog sound signal is digitized at a frequency Fs ′ smaller than the sampling frequency Fs, and a predetermined number N smaller than the predetermined number N An acoustic frame acquisition means for acquiring an acoustic frame composed of a sample of ', a frequency conversion means for obtaining a spectrum by performing frequency conversion on the acoustic frame using a window function, and a predetermined frequency from the generated spectrum. Extract multiple spectral sets in the frequency range, calculate the spectral intensity of each spectral set, and An encoding unit for extracting one or more embedded bit strings based on each spectrum intensity, and an additional information extracting unit for extracting the additional information by converting the output bit string according to a predetermined rule in units of words. An apparatus for extracting information from an acoustic signal is provided.

  According to the present invention, based on the restriction of the voice recording performance of the user's mobile terminal, digitization is performed at a frequency Fs ′ smaller than the sampling frequency Fs of the original acoustic signal, and the number is smaller than the number N of samples of one acoustic frame at the time of embedding. Since the sampling process is performed with the number of samples N ′ as one sound frame, sound signal input can be realized without any trouble even in a portable terminal having low voice recording performance. Thus, the extraction process is realized while maintaining the same accuracy, so that it is possible to perform the extraction process without delay even in a portable terminal having a low processing capability.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. First embodiment)
First, the first embodiment will be described. First, the change state of the predetermined frequency component of the acoustic frame before and after the embedding process in the first embodiment will be described. FIG. 1 shows a state of a predetermined frequency component according to this embodiment in units of one acoustic frame. In each acoustic frame shown in FIG. 1, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction. A shaded portion indicates a portion where a frequency component exists, and the darker the shade, the stronger the component strength. In FIG. 1, the frequency region is divided into three in the frequency direction of the vertical axis, but the second region from the top, that is, the frequency band from F1 to F2 is the frequency band to be changed, and the topmost part. That is, the frequency band above the frequency F2 and at the bottom, that is, less than F1, is a frequency band that is 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. 1A, the spectrum in the front part of the change target frequency band is expressed by SP1, and the spectrum in the rear part of the change target frequency band is expressed by SP3.

  In the present embodiment, when the code 1 is embedded, as shown in FIG. 1B, the rear component of the L-ch signal to be changed is removed, and a component equivalent to the removed component is R-ch. Is added to the signal. Further, the intensity of the component at the front of the frequency band to be changed of the L-ch signal is increased, and the intensity of each spectrum set at the front of the R-ch signal is reduced. This state is referred to as “state 1”. When embedding the code 2, as shown in FIG. 1 (c), the front component of the change target frequency band of the L-ch signal is removed, and a component equivalent to the removed component is converted to the R-ch signal. to add. Further, the intensity of the rear component of the frequency band to be changed of the L-ch signal is increased, and the intensity of each spectrum set at the rear of the R-ch signal is reduced. This state is referred to as “state 2”.

  In this embodiment, information is embedded by changing the frequency component into two states as shown in FIGS. Since there are two states, this corresponds to an information amount of 1 bit.

  In the present embodiment, the change target frequency bands F1 to F2 are set to either “1.7 kHz to 3.4 kHz” or “3.4 kHz to 6.8 kHz”. This is due to the following reasons.

  The human sound source localization sensation tends to increase in the treble part, but the energy of the source sound source has a characteristic that it becomes smaller as the treble becomes higher, especially when it exceeds the telephone line bandwidth (300 Hz to 3.4 kHz), it becomes only the harmonic component. As a result, the audible sound source localization variation is small. As a result of the experiment, the audible sound source localization mutation increases in the region of 400 Hz to 1.5 kHz, but tends to decrease when the frequency exceeds 1.5 kHz, and almost disappears when the frequency exceeds 4 kHz. This is because there is almost no sound component above 4 kHz, and the fundamental tone of the instrument sound exceeds the highest range, so that only the overtone component is obtained.

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

(1.1. Configuration of embedded device)
First, an information embedding device for an acoustic signal will be described. FIG. 2 is a functional block diagram showing a configuration of an information embedding device for an acoustic signal. In FIG. 2, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a frequency component changing means, 40 is a frequency inverse converting means, 50 is a modified acoustic frame output means, 60 is a storage means, and 61 is an acoustic signal storage section. 62 is an additional information storage unit, 63 is a modified acoustic signal storage unit, and 70 is a bit array creation means. Note that the apparatus shown in FIG. 2 can handle both stereo sound signals and monaural sound signals, but here, a case where processing is performed on stereo sound signals will be described.

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

  The storage means 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded with additional information, an additional information storage unit 62 that is configured as a bit array and stores additional information embedded in the stereo acoustic signal, and an additional information It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after information is embedded, and stores various information necessary for other processing. The bit array creation unit 70 extracts additional information from the additional information storage unit 62, adds a 1-bit error detection bit (parity bit) to each word of the additional information, and then further adds 8 bits according to a predetermined rule. It has a function to create a bit array to which is added. The additional information is information that should be added to the sound information and embedded, and includes attribute information such as a title and artist name, and other information other than the attribute information. In the present embodiment, since the ASCII code is used as the code format of the additional information, 7 bits are set as one word in the additional information, and the bit array generated by the bit array generating means 70 is inverted after adding an error detection bit. The 16 bits subjected to the additional processing are defined as one word. Each component shown in FIG. 2 is actually realized by mounting a dedicated program on hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

(1.2. Processing operation of embedded device)
Next, the processing operation of the information embedding device for the acoustic signal shown in FIG. 2 will be described. Here, a case where processing is performed on a stereo sound signal having two channels of L (left) and R (right) as sound signals will be described. The sound frame reading means 10 reads a predetermined number of samples as one sound frame from each of the left and right channels of the stereo sound signal stored in the sound signal storage unit 61. The number of samples of one sound frame read by the sound frame reading means 10 can be set as appropriate, but is desirably about 4096 samples when the sampling frequency is 44.1 kHz. Therefore, the acoustic frame reading means 10 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames.

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

  The frequency conversion unit 20 performs frequency conversion on the sound frame read by the sound frame reading unit 10 to obtain a frame spectrum that is a spectrum of the sound frame. Specifically, frequency conversion is performed using a window function. As frequency conversion, Fourier transform, wavelet transform, and other various known methods can be used. In the present embodiment, a case where Fourier transform is used will be described as an example.

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

  In this embodiment, not only to prevent the generation of the pseudo-harmonic component, but also to create a plurality of states for embedding information from one acoustic frame, a plurality of window functions are prepared, and one acoustic frame is provided. On the other hand, Fourier transform is performed using each window function to obtain a plurality of spectra. As a plurality of window functions, in the present embodiment, the first window function W (1, i), the second window function W (2, i), the third window as shown in FIGS. A function W (3, i) is prepared so that it can be easily recognized on the extraction side. The first window function W (1, i) is for extracting the front part of the acoustic frame, and has a maximum value of 1 at the position of a predetermined sample number i in the front part as shown in FIG. And the rear part 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 (1, i), but in this embodiment, it is defined by [Equation 1] described later. By multiplying the window function W (1, i), the signal waveform of the acoustic frame as shown in FIG. 3A has a signal component remaining in the front part as shown in FIG. The component is deleted, and this becomes a Fourier transform target.

  The second window function W (2, i) is for extracting the central portion of the acoustic frame, and as shown in FIG. 3C, at the position of the predetermined sample number i in the central portion. The maximum value is 1, and the minimum value is set to 0 at the front and rear portions. Which sample number takes the maximum value depends on the design of the window function W (2, i), but in this embodiment, it is defined by [Expression 2] described later. By multiplying the window function W (2, i), the signal waveform of the acoustic frame as shown in FIG. 3A has a signal component remaining in the center as shown in FIG. The rear signal component is removed, and this is subjected to Fourier transform.

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

  Thus, after extracting the front part, the central part, and the rear part, the Fourier transform is executed, so that spectra corresponding to the front part, the central part, and the rear part are obtained. In order to embed a bit value in one acoustic frame, the bit value is originally divided into two parts, a front part and a rear part. However, on the extraction side, it is not always possible to read a signal synchronously. Therefore, in order to clearly distinguish the front part and the rear part, in this embodiment, the signal component at the center part is always deleted at the time of embedding, and the front part and the rear part are separated in time (however, at the time of extraction) Only need to analyze the front and rear, and ignore the middle). In the window function used in the present embodiment, the window function W (1, i) and the window function W (3, i) are asymmetrical, so that erroneous recognition of embedded information is less likely to occur during extraction.

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

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

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

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

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

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

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

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

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

  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). Using the three window functions W (1, i), W (2, i), W (3, i), processing according to the following [Equation 5] is performed, and the conversion data corresponding to the left channel is obtained. Real part Al (1, j), Al (2, j), Al (3, j), imaginary part Bl (1, j), Bl (2, j), Bl (3, j), corresponding to the right channel Real part Ar (1, j), Ar (2, j), Ar (3, j), imaginary part Br (1, j), Br (2, j), Br (3, j) obtain. Note that the window functions W (1, i), W (2, i), and W (3, i) are functions whose values increase near the front (front), near the center, and near the rear of the acoustic frame, respectively. ing.

[Formula 5]
Al (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xl (i) .cos (2πij / N)
Bl (1, j) = Σi _{= 0,..., N-1} W (1, i) · Xl (i) · sin (2πij / N)
Al (2, j) = Σi _{= 0,..., N-1} W (2, i) · Xl (i) · cos (2πij / N)
Bl (2, j) = Σi _{= 0,..., N-1} W (2, i) · Xl (i) · sin (2πij / N)
Al (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xl (i) · cos (2πij / N)
Bl (3, j) = Σi _{= 0,..., N-1} W (3, i) .Xl (i) .sin (2πij / N)
Ar (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xr (i) .cos (2πij / N)
Br (1, j) = Σ _{i = 0,..., N-1} W (1, i) · Xr (i) · sin (2πij / N)
Ar (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i) .cos (2πij / N)
Br (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i) .sin (2πij / N)
Ar (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xr (i) · cos (2πij / N)
Br (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xr (i) · sin (2πij / N)

  When the frequency conversion means 20 performs Fourier transform on the B type sound frame, the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). The window function W (4, i) is used to perform processing according to the following [Equation 6], and the real part Al (4, j) and imaginary part Bl (4, j) of the conversion data corresponding to the left channel ), Real part Ar (4, j) and imaginary part Br (4, j) of the conversion data corresponding to the right channel are obtained.

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

  In the above [Formula 5] and [Formula 6], i is a serial number assigned to N samples in each acoustic frame, and takes an integer value of i = 0, 1, 2,. 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 5] and [Equation 6], a frame spectrum in which the signal component of each acoustic frame is represented by a spectrum corresponding to the frequency is obtained. Subsequently, the frequency component changing unit 30 extracts a spectrum set in a predetermined frequency range from the generated frame spectrum. In the present embodiment, a range between F1 and F2 is extracted.

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

  In the present invention, when the component intensity in the change target frequency band of the L-ch signal is increased and the intensity of the corresponding spectrum set of the R-ch signal is decreased, the window function F that changes according to the value j in the frequency direction. (J) is used. This frequency direction window function F (j) is defined by the following [Equation 7].

[Formula 7]
F (j) = 1.0− (j−Pr) ² / Pr ²

  In the above [Expression 7], Pr = P / 2. Further, P is the width in the frequency direction of the change target frequency band. When m is the number of the lower limit component of the change target frequency band and M is the number of the upper limit component of the change target frequency band, P = M−m It is. For example, when 1.7 kHz to 3.4 kHz is set as the change target frequency band, since m = 160 and M = 320, P = 160. A curve drawn by the window function F (j) is shown in FIG. As shown in FIG. 4, the window function F (j) is a function in which the weight is increased at the center portion of the frequency in the change target frequency band. When the window function F (j) is actually applied, the value of j is corrected by m which is the lower limit of the change target frequency band.

  When the information to be embedded is “value 1”, the state of the frequency component is set to “state 1”, that is, as shown in FIG. Change to state.

[Formula 8]
For each component of j = m to M−1, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = F (j−m) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = (1.0−F (j−m)) · Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m)) · Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  In the above [Formula 8], Al ′ (3, j) and Bl ′ (3, j) are both 0 at j = m to M−1. As shown in the upper part of FIG. 1B, this indicates that each component in SP3 is set to 0 in L-ch, but “state 1” makes the difference from SP1 clear. Therefore, it is not always necessary to set to 0, and a small value is sufficient.

  When the information to be embedded is “value 2”, the state according to the following [Equation 9] is executed to change the frequency component state to “state 2”, that is, as shown in FIG. Change to the correct state.

[Formula 9]
For each component of j = m to M−1, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (3, j) = F (j−m) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · E (3, j) / {Al (3, j) ² + B1 (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m)) · Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m)) · Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  In the above [Equation 9], in j = m to M−1, both Al ′ (1, j) and Bl ′ (1, j) are set to 0. This indicates that each component in SP1 is set to 0 in L-ch as shown in the upper part of FIG. 1C, but “state 2” makes the difference from SP3 clear. Therefore, it is not always necessary to set to 0, and a small value is sufficient.

  As described above, since it is necessary to change the frequency component for the A type sound frame in accordance with the bit value to be embedded, the frequency component changing means 30 performs the above [Formula 8] and [Formula 9]. Execute the process according to. However, since the B type acoustic frame is used to prevent discontinuity at both end portions that occurs only in the case of the A type acoustic frame, it is necessary to change the frequency component according to the bit value. Absent. Therefore, the frequency component changing unit 30 executes processing according to the following [Formula 10] for the B type acoustic frame, and always removes the component of the frequency band to be changed.

[Formula 10]
For each component of j = 0 to M−1, Al ′ (4, j) = 0
Bl ′ (4, j) = 0
E (4, j) = {Al (4, j) ² + Bl (4, j) ² + Ar (4, j) ² + Br (4, j) ² } ^1/2
Ar ′ (4, j) = Ar (4, j) · E (4, j) / {Ar (4, j) ² + Br (4, j) ² } ^1/2
Br ′ (4, j) = Br (4, j) · E (4, j) / {Ar (4, j) ² + Br (4, j) ² } ^1/2

  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).

[Formula 11]
Xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) -Σ j Bl' (1, j) · sin (2πij / N)} + 1 / N · { Σ _j Al ′ (2, j) · cos (2πij / N) −Σ _j Bl ′ (2, j) · sin (2πij / N)} + 1 / N · {Σ _j Al ′ (3, j) · cos (2πij / N) −Σ _j Bl ′ (3, j) · sin (2πij / N)} + Xlp (i + N / 2)
Xr' (i) = 1 / N · {Σ j Ar' (1, j) · cos (2πij / N) -Σ j Br' (1, j) · sin (2πij / N)} + 1 / N · { _{Σ j Ar' (2, j)} · cos (2πij / N) -Σ j Br' (2, j) · sin (2πij / N)} + 1 / N · {Σ j Ar' (3, j) · cos _{(2πij / N) -Σ j Br'} (3, j) · sin (2πij / N)} + Xrp (i + N / 2)

In the above [Expression 11], 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 [Equation 11] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 11], 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 real part Al ′ (4, j) and imaginary part Bl ′ (4, j) of the left channel of the spectrum obtained by the above [Equation 10]. , Using the real part Ar ′ (4, j) and imaginary part Br ′ (4, j) of the right channel, the processing according to the following [Equation 12] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For frequency components that are not modified in the above [Equation 10], in the following [Equation 12], Al ′ (4, j), Bl ′ (4, j), Ar ′ (4, j), Br The original values Al (4, j), Bl (4, j), Ar (4, j), and Br (4, j) are used as ′ (4, j).

[Formula 12]
Xl' (i) = 1 / N · {Σ j Al' (4, j) · cos (2πij / N) -Σ j Bl' (4, j) · sin (2πij / N)} + Xlp (i + N / 2 )
Xr' (i) = 1 / N · {Σ j Ar' (4, j) · cos (2πij / N) -Σ j Br' (4, j) · sin (2πij / N)} + Xrp (i + N / 2 )

  By the above [Equation 12], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified sound frame are obtained.

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

  Next, the overall flow of processing of the information embedding device for the acoustic signal shown in FIG. 2 will be described with reference to the flowchart of FIG. Each component which comprises the apparatus shown in FIG. 2 cooperates, and performs the process according to FIG. FIG. 5 corresponds to processing for one word of additional information. Although one word can be set to an arbitrary number of bits, as described above, in this embodiment, it is set to 1 byte (8 bits) including error detection bits. Further, since information embedding is performed on an A type acoustic frame, FIG. 5 illustrates the A type acoustic frame. The B type acoustic frame is read by the acoustic frame reading means 10 in parallel with the A type acoustic frame, and is frequency converted by the frequency converting means 20 using the window function W (4, i). The frequency component changing unit 30 removes the component of the change target frequency band, the frequency inverse converting unit 40 performs the frequency inverse conversion, and then the modified acoustic frame output unit 50 outputs the result.

  In FIG. 5, first, the bit array creation means 70 performs a predetermined process on the additional information extracted from the additional information storage unit 62 to create a new bit array (S101). Specifically, first, it is extracted in units of 1 word (7 bits) from the additional information storage unit 62, and 1 bit of error detection bits is added to this to make 8 bits. Then, the extracted bits of one word (7 bits) are inverted and arranged, and finally, one bit having the same value as the error detection bit is arranged to create a 16-bit bit array. FIG. 6 shows how the bit arrangement is changed by this processing. FIG. 6A shows the bit arrangement after adding the error detection bits. In FIG. 6A, the eighth bit “0” is an error detection bit. In such a case, “01100001” obtained by inverting “1011110” of the first to seventh bits is changed to the ninth bit to the fifteenth bit, and “0” of the eighth bit is copied to become the sixteenth bit. Create a bit array. As a result, the arrangement as shown in FIG.

  In the arrangement shown in FIG. 6B, the order of the corresponding bits in the first 8 bits and the latter 8 bits is the same. Therefore, at the time of extraction, any bit is compared with a bit 8 bits away. Will do. That is, as shown in FIG. 6B, the first bit is compared with the ninth bit (first bit inversion), and the second bit is compared with the tenth bit (second bit inversion). In this case, at the time of extraction, even if the second bit is erroneously determined to be the head, it is compared with the tenth bit, and compatibility is often maintained. Then, there is a possibility of extracting an incorrect bit arrangement starting from the second bit. In order to prevent such a problem from occurring, in the present embodiment, the odd-numbered bits and the even-numbered bits are further interchanged in the latter 8 bits of 16 bits. As a result, the arrangement as shown in FIG. These 16 bits are read into a register in a computer used as an information embedding device for an acoustic signal. Thus, in the additional information storage unit 62, one word is 7 bits, but at the time of embedding processing, processing for one word in the additional information is performed with this 16-bit array. In the bit array thus created, the first half bit string is referred to as the first half bit string, and the second half bit string is referred to as the second half bit string.

  Next, the frequency component changing unit 30 performs a process of reading 1 bit from 16 bits held in the register (S102). Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as one acoustic frame from each of the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S103). Subsequently, the frequency converting unit 20 and the frequency component changing unit 30 perform processing for changing the state of the frequency component of the acoustic frame to either “state 1” or “state 2” (S104). Specifically, 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 5] is performed using three window functions W (1, i), W (2, i), and W (3, i).

  Subsequently, the frequency component changing unit 30 executes processing according to the above [Equation 8] and [Equation 9] according to the value 1 and the value 2 received from the bit array creation unit 70, and the component of the frequency band to be changed The state is changed to either “state 1” or “state 2”.

  Next, the frequency inverse transform means 40 performs the process of obtaining the modified acoustic frame by performing the frequency inverse transform on the frame spectrum in which the ratio between the spectrum sets of each window component is changed by the process of S104 (S105). As a matter of course, this frequency inverse transform needs to correspond to the method executed by the frequency converter 20 in S104. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the 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 11] 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. When the processing for one acoustic frame is completed 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), the process ends. When the processing corresponding to each bit of the bit arrangement (16 bits) for one word read in S101 is completed, the processing returns from S102 to S101, and processing for creating the bit arrangement by reading the next word of the additional information. Will do. When the processing is completed for all the words of the additional information, the processing returns to the first word of the additional information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

  In the present embodiment, the case has been described where the additional information is 7 bits per word, 1 bit for error detection is added, and then the inverted bit is added to 16 bits to process 1 word of additional information. However, as long as there is an agreement with the extraction side, one word of additional information can be recorded in an arbitrary number of bits.

  Of the left channel of the modified acoustic signal obtained as described above, with respect to the portion where the additional information is embedded, the component of the change target frequency band has only two distributions of the state 1 and the state 2 described above. become. However, since the components other than the component of the frequency band to be changed remain as the original acoustic signals, various distributions are made based on the setting of the producer. Further, as shown in the above example, when a stereo sound signal is used, the component of the frequency band to be changed that has been changed in the left channel is also apparent from the processing of [Equation 8] and [Equation 9]. In this way, it is always added to the component of the frequency band to be changed in the right channel. Therefore, since the right channel supplements the deleted component in the left channel, there is no signal degradation when viewed as both channels as a whole.

(1.3. Apparatus for extracting information from acoustic signals)
Next, an apparatus for extracting information from an acoustic signal according to the present invention will be described. FIG. 7 is a block diagram showing an embodiment of an apparatus for extracting information from an acoustic signal according to the present invention. In FIG. 7, 100 is an acoustic signal input means, 110 is a reference frame acquisition means, 120 is a phase change frame setting means, 130 is a frequency conversion means, 140 is a code determination parameter calculation means, 150 is a code output means, and 160 is additional information. Extraction means 170 is an acoustic frame holding means.

  The acoustic signal input unit 100 has a function of acquiring and inputting a flowing sound as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. Any microphone can be used, regardless of whether it is monaural omnidirectional or stereo directional, as long as the component in the frequency band to be changed can be detected. Even if it is stereo-directional, only one channel needs to be used. In addition, when information is embedded by the apparatus shown in FIG. 2, the information can be extracted even if a microphone having a general accuracy is used instead of a high accuracy one. The reference frame acquisition unit 110 has a function of reading an audio frame composed of a predetermined number of samples as a reference frame from the input digital monaural audio signal (or one channel of a stereo audio signal). The phase change frame setting means 120 has a function of setting, as a phase change frame, an acoustic frame whose phase has been changed by moving the reference frame and a predetermined sample at a time. The frequency conversion means 130 has the same function as the frequency conversion means 20 shown in FIG.

The code determination parameter calculation means 140 extracts each frequency intensity data corresponding to a predetermined frequency range from the generated frame spectrum, and sets the intensity values E _C1 and E _C2 of each frequency intensity data corresponding to each component as follows: Formula 13] is calculated, the intensity values E _C1 and E _C2 are used as code determination parameters, and a function for determining a predetermined state based on the ratio of the code determination parameters E _C1 and E _C2 is provided. is doing.

[Formula 13]
E _C1 = Σ _{j = m,..., M−1} {Al (1, j) ² + Bl (1, j) ² } · F (j−m) · C
E _C2 = Σ _{j = m,..., M−1} {Al (3, j) ² + Bl (3, j) ² } · F (j−m) · C

  The coefficient C in the above [Formula 13] is defined by the following [Formula 14].

[Formula 14]
C = 1.0 / Σ _{j = 0, P-1} F (j)

  The code output means 150 determines what is determined to be the optimum phase from the acoustic frames (reference frame and phase change frame) corresponding to one reference frame, and selects a code corresponding to the state of the acoustic frame. It has a function to output. The code determination parameter calculation unit 140 and the code output unit 150 constitute an encoding unit. The additional information extraction unit 160 has a function of converting the binary array output by the code output unit 150 according to a predetermined rule and extracting it as meaningful additional information. The acoustic frame holding means 170 is a buffer memory that can hold two consecutive reference frames. Each component shown in FIG. 7 is actually realized by mounting a dedicated program on hardware such as a small computer having an information processing function and its peripheral devices. In particular, in the present invention, it is assumed that a portable game machine, a PDA, a PHS terminal, and a mobile phone, which have lower recording performance and arithmetic processing performance than a general-purpose computer, are used as the extraction device.

(1.4. Processing operation of extraction device)
Next, the processing operation of the apparatus for extracting information from the acoustic signal shown in FIG. 7 will be described. The extraction device according to the present invention can be set not to perform error correction when an error is detected by a parity check, or can be set to perform 1-bit error correction. Hereafter, the processing operation of the extraction apparatus in a setting in which error correction is not performed will be described according to the flowchart of FIG. First, in this apparatus, the phase determination table S (p), the phase determination log, the phase determination flag, and the bit counter are initialized (S200). The phase determination table S (p) is a table for determining the phase, and p takes an integer value of 0 to 5. The initial value is set to S (p) = 0. The phase determination log records the determined phase, that is, the phase number p for each set of one reference frame and five phase change frames, and 0 is set in the initial state. The phase determination flag is a flag indicating whether or not the phase is fixed, and is set to Off in the initial state. For the bit counter, 0 is set as an initial value.

  In this way, when the initial value is set and the user wants to know the attribute information such as the song name of the music that is flowing, first, the extraction device Instruct startup. For example, this can be executed by operating a predetermined button when the extraction device is realized by a mobile terminal such as a mobile phone. When an instruction is input to the extraction device, the acoustic signal input unit 100 records the flowing music, digitizes it, and inputs it as a digital acoustic signal. More specifically, the audio input from the omnidirectional microphone (or one channel of the directional microphone) is digitized by the A / D converter. At this time, the sampling frequency and the number of quantization bits depend on the performance of the device. However, in this embodiment, since a portable game machine, a PDA, and a PHS terminal are assumed, the sampling frequency is 22.05 kHz and the number of quantization bits. Digitize with 12 bits.

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 100 as a reference frame (S201). Specifically, the reference frame is extracted and read into the acoustic frame holding unit 170. The number of samples of one sound frame read as the reference frame by the reference frame acquisition unit 110 is set to 2048, which is half of that set by the sound frame reading unit 10 shown in FIG. This is because the sampling frequency Fs ′ by the acoustic signal input means 100 is 22.05 kHz, which is half the sampling frequency Fs (44.1 kHz) of the original acoustic signal processed by the embedding device. That is, in the extraction device, one acoustic frame 2048 (N) is used so that one acoustic frame in the embedding device having a sampling frequency of 44.1 kHz and the number of samples 4096 (N) is equal to the number of seconds of one acoustic frame in the extraction device. '= (N / 2)) sample is set. That is, N ′ is set so that N ′ = N × Fs ′ / Fs. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 2048 samples as reference frames. In the present embodiment, when sampling is performed with the quantization signal number 12 bits in the acoustic signal input means 100, the reference frame acquisition means 110 performs extraction after up-converting each sample to 16 bits.

  The acoustic frame holding means 170 can store two reference frames as described above, and when a new reference frame is read, the old reference frame is discarded. Therefore, the sound frame holding means 170 always stores two reference frames (continuous 4096 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. For the reference frame, after setting sample number 1 to sample number 2048 as the first reference frame, the next reference frame is sample number 2049 to sample number 4096, and the next reference frame is sample number 4097 to sample number 6144. It is set without interruption. Then, for each reference frame, five phase change frames moved by 1/6 frame (341 samples) are set. For example, for the first reference frame, five phase change frames configured by 2048 samples starting from sample numbers 342, 684, 1025, 1366, and 1708 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 the 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 is initialized (S303). The candidate code table records a phase number of 0 to 5 that specifies one reference frame and five phase change frames, and a binary code obtained from the states of the six acoustic frames.

  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 a frame spectrum (S401). Specifically, the processing according to [Formula 5] is executed in the same manner as the frequency conversion means 20 shown in FIG. However, N is replaced by N ′ (= N / 2) in [Formula 5]. Further, W (1, i) to W (3, i) used in [Formula 5] are obtained by replacing N in [Formula 1] to [Formula 3] with N ′ (= N / 2), respectively. . Since only the left channel is used for extraction, the processing according to [Formula 5] is performed, and the imaginary part Bl (1) such as the real part Al (1, j) of the converted data corresponding to the left channel is performed. , J) etc.

By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, the intensity value E _C1 and the intensity value E _C2 are used to determine what state the component of the frequency band to be changed is, that is, what value was embedded as a 1-bit value. A determination process is performed (S402). Specifically, first, the following determination processing is executed, and as a result, it is determined that the corresponding state is obtained, and the corresponding value is output.

If E _C1 > E _C2 , it is determined that the state is “1”, and a value of 1 is output.
When E _C2 ≧ E _C1 , it is determined that “state 2” is satisfied, and a value 2 is output.

  The code determination parameter calculation unit 140 sets either value 1 or value 2 according to the determination result for each acoustic frame (S403). If either of the values 1 and 2 is set as a result of the determination, the phase determination table S (p) is further updated according to the following [Equation 15] (S404).

[Formula 15]
When it is determined that the state is “1” and the value 1 is output, S (p) ← S (p) + E _C1 / E _C2
When it is determined that the state is “2” and the value 2 is output, S (p) ← S (p) + E _C2 / E _C1

Here, returning to the flowchart of FIG. 9, the code determination parameter calculation unit 140 stores a candidate for the optimum phase in the candidate code table (S304). Specifically, the value of the phase number p that maximizes the value of S (p) recorded in the phase determination table, the binary code set in S403, and the above [ One of E _C1 and E _C2 corresponding to the frequency component calculated by executing the processing according to Equation 13] is stored in the candidate code table as an optimal phase candidate.

  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 p recorded in the candidate storage table is the optimum phase, and the code recorded in the candidate storage table 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 phase determination flag is always On, so that the phase determination processing (S303 to S309) is not performed and only the code determination processing (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 15 or less or 16 or more (S205). If the bit counter is equal to or smaller than 15, the process returns to S201 to perform processing for extracting the next reference frame.

  When the bit counter is 16 or more, the compatibility of the first half 8 bits and the second half 8 bits in the bit arrangement stored in the buffer is determined (S206). That is, as shown in FIG. 6C, the first and tenth bits, the second and ninth bits, the third and twelfth bits, the fourth and eleventh bits, The 5th and 14th bits, the 6th and 13th bits, the 7th and 16th bits are inverted from each other, and it is determined whether the 8th and 15th bits match. . If these conditions are satisfied, it is determined to be conforming, and if they are not satisfied, it is determined to be nonconforming. If the result of determination is incompatibility, processing returns to S201 and the next reference frame is extracted. As a result of the determination, if it matches, a parity check is performed on the first 8 bits of 16 bits (S207). Specifically, 8 bits in the first half of 16 bits are extracted, parity calculation is performed on the previous 7 bits, and the result is compared with the 8th bit. If the parity check fails, the process returns to S201 to perform processing for extracting the next reference frame.

  If the parity check is successful, the additional information extraction unit 160 adds 1 bit to the previous 7 bits in the first 8 bits and outputs the result (S208). Here, if the parity check passes, there is a high possibility that the eighth bit used for collation is an error detection bit. Then, the previous 7 bits are considered to be 7 bits in the original additional information. Therefore, by adding bit 0 to the 7 bits from the head, it is output as one word in the ASCII code. On the other hand, if the parity check fails, there is a high possibility that the eighth bit used for collation is not an error detection bit. Then, it is considered that the 7 bits held at that time are shifted from the 7 bits in the original additional information. In this case, the first 1 bit is discarded, and processing for obtaining a new 1 bit obtained by the processing from S201 to S204 is performed.

  If the parity check is performed in this way and the result is passed, there is a high possibility that the part is a word delimiter. Extraction can be done in units. On the other hand, even if the parity check is passed, it may be a coincidence and not actually a word break. In such a case, there is a high possibility that it will be rejected at the next parity check, and the correct delimiter can be accurately determined after repeating several times. In S208, when 1 bit is added to the previous 7 bits and output, the bit counter is initialized to 0 (S209). Then, the process returns to S201 to perform processing for extracting the next reference frame.

  By executing the processing shown in FIG. 8 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. In this embodiment, the ASCII code is adopted. . That is, the additional information extraction unit 160 recognizes the bit value array obtained from the code determined by the code determination parameter calculation unit 140 and output from the code output unit 150 in units of 1 byte (8 bits), and recognizes this as ASCII. Recognizes character information according to the code. The character information thus obtained is displayed and output on a screen of a display device (not shown).

  Therefore, if the embedding device embeds the attribute information such as the song title or artist in the sound signal as the character information, the user wants to know the song title or artist by listening to the music being played. If a predetermined operation is performed on the mobile terminal that functions as the extraction device, attribute information such as a song title and an artist is displayed as character information on the screen of the mobile terminal.

  In the above example, the processing operation of the extraction apparatus in a setting in which error correction is not performed has been described according to the flowchart of FIG. When error correction is not performed, it is possible to output one word of additional information only when no error occurs. For example, in the case of a normal array as shown in FIG. 12 (a) (same as the array in FIG. 6 (c)), the first bit to the seventh bit are compared with the respective inverted bits, and the eighth bit Verifies the copy bit. If the relationship is suitable, a parity check is performed on the first to eighth bits of the first half. In the example of FIG. 12A, the first bit to the eighth bit and the ninth bit to the sixteenth bit are suitable, and the first half to the eighth bit parity check is also passed. By adding bit 0 to bit-7, it is output as one word in the ASCII code.

  Next, the processing operation of the extraction apparatus in a setting for performing 1-bit error correction will be described with reference to the flowchart of FIG. 11 includes a portion that performs the same processing as in FIG. Therefore, portions that perform the same processing as in FIG. Also in the example of FIG. 11, first, initialization processing is performed (S200). In this initialization process, as in the example of FIG. 8, 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. 8, after extracting an acoustic frame composed of a predetermined number of samples as a reference frame (S201), after identifying the phase of each read acoustic frame, the embedded information is determined. The corresponding code is output (S202). Note that the processing of S202 is as shown in FIGS.

  When the phase is determined and the code is output, 1 bit corresponding to the output code value is stored in the buffer (S203), and the bit counter is incremented by "1" (S204). Then, it is determined whether the bit counter is 15 or less or 16 or more (S205). If the bit counter is 15 or less, the process returns to S201 to perform processing for extracting the next reference frame.

  When the bit counter is 16 or more, the compatibility of the first half 8 bits and the second half 8 bits in the bit string stored in the buffer is determined (S206). As a result of the determination, if it is not suitable for 2 bits or more, after the automatic correction mode is set to OFF (S210), the process returns to S201 to extract the next reference frame.

  On the other hand, if the result of determination is nonconformity of 1 bit or less, it is further determined whether it is compatible or only 1 bit is nonconforming. If it matches, parity check is performed on the first 8 bits of 16 bits (S207). If the parity check is acceptable, the automatic correction mode is set to ON (S211), and then the additional information extraction unit 160 adds 1 bit to the previous 7 bits in the first 8 bits and outputs (S208).

  If only one bit is unsuitable, it is determined whether the automatic correction mode is ON or OFF (S212). If the automatic correction mode is OFF, the process returns to S201 to perform processing for extracting the next reference frame. If the automatic correction mode is ON, a parity check is performed on the first 8 bits of 16 bits (S207). If the parity check is successful, the additional information extraction unit 160 adds 1 bit to the previous 7 bits in the first 8 bits and outputs the result (S208). If the parity check fails, the first non-conforming bit is inverted and corrected (S213). Then, the additional information extraction unit 160 adds 1 bit to the previous 7 bits in the first 8 bits and outputs the result (S208).

  In S208, when 1 bit is added to the previous 7 bits and output, the bit counter is initialized to 0 (S209). Then, the process returns to S201 to perform processing for extracting the next reference frame.

  As in the example of FIG. 11, in the case of setting for 1-bit error correction, even if a 1-bit error occurs, it is possible to output 1 word of additional information. For example, consider the case where the fifth bit inversion is an error, as shown in FIG. In this case, the fifth bit and the fifth bit inversion have the same value and are not in an inversion relationship, which is incompatible. However, since the parity check in the first half is passed, one bit is added to the first 7 bits out of the 8 bits in the first half and output.

  Next, consider the case where the fifth bit is in error as shown in FIG. In this case, the fifth bit and the fifth bit inversion have the same value and are not in an inversion relationship, which is incompatible. Furthermore, since the parity check in the first half is rejected, the fifth bit which is the non-conforming bit in the first half is inverted and corrected, and then 1 bit is added to the previous 7 bits out of the 8 bits after the correction.

  Next, as shown in FIG. 13B, consider the case where the eighth bit copy has an error. In this case, the 8th bit and the 8th bit copy are different values and are not in a relationship of copying, which is incompatible. However, since the parity check in the first half is passed, one bit is added to the first 7 bits out of the 8 bits in the first half and output.

  Next, consider the case where the eighth bit is in error as shown in FIG. In this case, the 8th bit and the 8th bit copy are different values and are not in a relationship of copying, which is incompatible. Further, since the parity check in the first half is rejected, the eighth bit, which is a non-conforming bit in the first half, is inverted and corrected, and then 1 bit is added to the previous 7 bits out of the 8 bits after the correction.

  When the odd-numbered bits and the even-numbered bits are interchanged in the latter-half bit string, as shown in FIG. 6C, the odd-numbered bits in the first-half bit string are compared with the bits separated by 9 bits, and the even-numbered bits in the first-half bit string The th bit is compared with a bit 7 bits away. That is, the first bit is compared with the tenth bit (first bit inversion), and the second bit is compared with the ninth bit (second bit inversion). In this case, at the time of extraction, if the second bit is erroneously determined to be the head, it is compared with the eleventh bit. Since this is the fourth bit inversion, compatibility is often not maintained. Then, the possibility of extracting an incorrect bit arrangement starting from the second bit is reduced. For this reason, when the odd-numbered bits and the even-numbered bits are interchanged in the second half bit string, compared to the case where the order of the corresponding bits in the first half bit string and the second half bit string is the same as shown in FIG. The recognition is low and it becomes possible to perform extraction with higher accuracy.

(1.5. About phase correction processing)
As described above, at the time of extraction, it is not always possible to read an acoustic signal corresponding to the acoustic frame embedded at the time of embedding. Therefore, the phase of the acoustic frame is shifted and read in a plurality of ways (six in this embodiment), the optimum phase is determined, and a code corresponding to the acoustic frame specified by the phase is output. Yes. For example, when reading in six ways, the first sound frame is originally a sample of sample numbers 1 to 2048, but six pieces composed of 2048 samples starting from sample numbers 1, 342, 684, 1025, 1366, and 1708 Are processed, and a code corresponding to the optimum acoustic frame is output. As described with reference to the flowchart of FIG. 9, in the present embodiment, when the same phase continues a predetermined number of times, the processing is performed after that phase is determined.

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

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

  In this case, the information embedding process in the embedding apparatus shown in FIG. 2 is also performed according to the flowchart of FIG.

  Therefore, as a process for setting the state 1 and the state 2 in S104, first, the fixed value V calculated according to the following [Equation 16] is set as the intensity of the component of the change target frequency band.

[Formula 16]
V = {0.5 · Lev · C / P} ^1/2

  In the case of the state 1, after executing the process according to the above [Equation 8], the process according to the following [Equation 17] is executed.

[Formula 17]
For each component of j = m to M−1, Al ′ (1, j) = F (j−m) · Al (1, j) · V / {Al (1, j) ² + Bl (1, j ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · V / {Al (1, j) ² + B1 (1, j) ² } ^1/2

  In the case of the state 2, after executing the process according to the above [Expression 9], the process according to the following [Expression 18] is executed.

[Formula 18]
For each component of j = m to M−1, Al ′ (3, j) = F (j−m) · Al (3, j) · V / {Al (3, j) ² + Bl (3, j ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · V / {Al (3, j) ² + B1 (3, j) ² } ^1/2

  As described above, even when information is embedded when the frequency component is small, the configuration of the extraction device for extracting information from the acoustic signal on the extraction side is the same as that in FIG. 7, and the processing operation is shown in FIGS. 11 is the same as that according to the flowchart of FIG.

(2. Second Embodiment)
Next, a second embodiment will be described. As shown in FIG. 15, the second embodiment is characterized in that the components of the frequency band to be changed in the acoustic frame are changed to four states and 2-bit information is embedded.

  Also in each acoustic frame shown in FIG. 15, the horizontal axis indicates the time direction and the vertical axis indicates the frequency direction, as in FIG. A shaded portion indicates a portion where a frequency component exists, and the darker the shade, the stronger the component strength. Also in FIG. 15, the predetermined frequency range (change target frequency band) is set to F1 or more and F2 or less as in FIG. 1, but in FIG. 15, the change target frequency band is two in the middle of F1 and F2. It is divided. Therefore, as shown in FIG. 15, the frequency domain is divided into four in the frequency direction of the vertical axis.

  In the present embodiment, when embedding reference numerals 1 and 2, as shown in FIGS. 15B and 15C, processing is performed in the same manner as in the first embodiment shown in FIGS. Is called.

  When the code 3 is embedded, as shown in FIG. 15D, the front upper frequency component and the rear lower frequency component of the L-ch signal are removed, and a component equivalent to the removed component is removed from the R-ch. Add to signal. Further, the intensity of the lower frequency component and rear upper frequency component of the change target frequency band of the L-ch signal is increased, and the intensity of each spectrum set corresponding to the R-ch signal is decreased. This state is referred to as “state 3”. When embedding the code 4, as shown in FIG. 15 (e), the lower frequency component in the front part and the higher frequency component in the rear part of the frequency band to be changed of the L-ch signal are removed, and it is equivalent to the removed component. The component is added to the R-ch signal. Further, the intensity of the upper frequency component in the front part and the lower frequency component in the rear part of the change target frequency band of the L-ch signal is increased, and the intensity of each spectrum set corresponding to the R-ch signal is weakened. This state is called “state 4”.

  In the present invention, information is embedded by changing the components of the frequency band to be changed to four states as shown in FIGS. Since four states can be embedded, this corresponds to an information amount of 2 bits.

(2.1. Embedded device)
Also in the second embodiment, the configuration of the embedding device is the same as that shown in FIG. 2, but the contents of the frequency component changing means 30 are mainly different. Specifically, the frequency component changing unit 30 performs a process for changing the ratio of the components of the change target frequency band according to the bit arrangement created by the bit arrangement creating unit 70 for the A type acoustic frame. In the second embodiment, the bit array is read two bits at a time, and 2-bit information is embedded in one acoustic frame. There are four embedded 2-bit values: “00”, “01”, “10”, and “11”. In the present embodiment, these are defined as value 1 to value 4. At this time, any one of four values “00” to “11” may be defined as a value 1 to a value 4. This is because the extraction side only needs to be able to specify the 2-bit array embedded on the embedding side. Therefore, this definition must match between the embedding side and the extraction side.

  Then, the frequency component changing unit 30 changes the frequency component to any one of the state 1 to the state 4 according to the value 1 to the value 4.

  When the two bits to be embedded are “value 1”, the processing according to the following [Equation 19] is executed to change the component state of the frequency band to be changed to “state 1”, that is, FIG. Change to the state shown in.

[Formula 19]
For each component of j = m to m + P−1 and m + P to M−1, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
For each component of j = m to m + P−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = F (j−m) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = (1.0−F (j−m)) · Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m)) · Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
For each component of j = m + P to M−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = F (j−m−P) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = F (j−m−P) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = (1.0−F (j−m−P)) · Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m−P)) · Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j ² } ^1/2
For each component of j = m to M−1, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  In the above [Equation 19], in j = m to m + P−1 and m + P to M−1, both Al ′ (3, j) and Bl ′ (3, j) are 0. This indicates that each component in SP3U and SP3D is set to 0 in L-ch as shown in the upper part of FIG. 15B, but “state 1” indicates the difference from SP1U and SP1D. Since it is sufficient if it can be clarified, it is not always necessary to set it to 0, and a small value is sufficient. When the two bits to be embedded are “value 2”, the processing according to the following [Equation 20] is executed to change the state of the component of the frequency band to be changed to “state 2”, that is, FIG. Change to the state shown in

[Formula 20]
For each component of j = m to m + P−1 and m + P to M−1, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
For each component of j = m to m + P-1, E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (3, j) = F (j−m) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · E (3, j) / {Al (3, j) ² + B1 (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m)) · Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m)) · Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
For each component of j = m + P to M−1, E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (3, j) = F (j−m−P) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m−P) · B1 (3, j) · E (3, j) / {Al (3, j) ² + B1 (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m−P)) · Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m−P)) · Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j ² } ^1/2
For each component of j = m to M−1, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  In the above [Expression 20], Al ′ (1, j) and Bl ′ (1, j) are both 0 in j = m to m + P−1 and m + P to M−1. This indicates that each component in SP1U and SP1D is set to 0 in L-ch as shown in the upper part of FIG. 15C, but “state 2” indicates a difference from SP3U and SP3D. Since it is sufficient if it can be clarified, it is not always necessary to set it to 0, and a small value is sufficient. When 2 bits to be embedded are “value 3”, the state of the component of the frequency band to be changed is changed to “state 3”, that is, FIG. Change to the state shown in.

[Formula 21]
For each component of j = m to m + P−1, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
For each component of j = m to m + P−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = F (j−m) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = (1.0−F (j−m)) · Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m)) · Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
For each component of j = m + P to M−1, E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (3, j) = F (j−m−P) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m−P) · B1 (3, j) · E (3, j) / {Al (3, j) ² + B1 (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m−P)) · Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m−P)) · Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j ² } ^1/2
For each component of j = m to M−1, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  In the above [Equation 21], when j = m to m + P−1, Al ′ (3, j) and Bl ′ (3, j) are both 0, and when j = m + P to M−1, Al ′ (1, j ), Bl ′ (1, j) are both 0. This indicates that each component in SP3D and SP1U is set to 0 in L-ch as shown in the upper part of FIG. 15D, but “state 3” indicates the difference from SP3U and SP1D. Since it is sufficient if it can be clarified, it is not always necessary to set it to 0, and a small value is sufficient. When 2 bits to be embedded are “value 4”, the state of the component of the frequency band to be changed is changed to “state 4”, that is, FIG. Change to the state shown in.

[Formula 22]
For each component of j = m to m + P−1, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
For each component of j = m to m + P-1, E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Al ′ (3, j) = F (j−m) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · E (3, j) / {Al (3, j) ² + B1 (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m)) · Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m)) · Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
For each component of j = m + P to M−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = F (j−m−P) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = F (j−m−P) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = (1.0−F (j−m−P)) · Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m−P)) · Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j ² } ^1/2
For each component of j = m to M−1, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  In the above [Equation 22], when j = m to m + P−1, both Al ′ (1, j) and Bl ′ (1, j) are set to 0, and when j = m + P to M−1, Al ′ (3, j ), Bl ′ (3, j) are both 0. This indicates that each component in SP1D and SP3U is set to 0 in L-ch as shown in the upper part of FIG. 15E, but “state 4” indicates a difference from SP3D and SP1U. Since it is sufficient if it can be clarified, it is not always necessary to set it to 0, and a small value is sufficient.

  Also in the second embodiment, as in the first embodiment, the frequency component changing unit 30 executes the process according to the above [Formula 10] for the B type acoustic frame, and always changes the frequency to be changed. Remove strip components. When the processing as described above is performed by the frequency component changing unit 30, the frequency inverse converting unit 40 and the modified acoustic frame output unit 50 are subjected to the same processing as in the first embodiment.

  The overall flow of the processing of the information embedding device for the acoustic signal in the second embodiment is basically the same as that in the first embodiment, but in the flowchart of FIG. 5, reading from the register in S102 is performed. Is in units of 2 bits, and is different in that it is set to any one of states 1 to 4 in S104.

(2.2. Extraction device)
Also in the second embodiment, the configuration of the extraction apparatus is the same as that shown in FIG. 7, but the contents of the sign determination parameter calculation means 140 are mainly different. Specifically, the code determination parameter calculation means 140 extracts each frequency intensity data corresponding to a predetermined frequency range from the generated frame spectrum, and the intensity values E _{C1 to} E _{C4 of} each frequency intensity data corresponding to each component. Is calculated based on the following [Equation 23], the intensity values E _{C1 to} E _C4 are used as code determination parameters, and it is determined that the state is a predetermined state based on the ratio of the code determination parameters E _{C1 to} E _C4. It has a function to do.

[Formula 23]
E _C1 = Σ _{j = m,..., M + P-1} {Al (1, j) ² + Bl (1, j) ² } · F (j−m) · C
E _C2 = Σ _{j = m,..., M + P-1} {Al (3, j) ² + Bl (3, j) ² } · F (j−m) · C
E _C3 = Σ _{j = m + P,..., M−1} {Al (1, j) ² + Bl (1, j) ² } · F (j−m−P) · C
E _C4 = Σ _{j = m + P,..., M−1} {Al (3, j) ² + Bl (3, j) ² } · F (j−m−P) · C

  The overall processing flow of the apparatus for extracting information from an acoustic signal in the second embodiment is basically the same as that in the first embodiment, but in the flowchart of FIG. The difference is that the value 4 is output and that 2 is added to the bit counter in S206.

  Further, the phase determination and the code output in step S202 are performed according to the flowcharts of FIGS. 9 and 10 as in the first embodiment. However, the processing of S402 to S404 in FIG. 10 is different from that of the first embodiment, and will be described below.

In the frequency component state determination of FIG. 10, the code determination parameter calculation means 140 uses the intensity value E _C1 to the intensity value E _C4 to determine what the frequency component state is, that is, a 2-bit state. A process of determining what value is embedded as a value is performed (S402). Specifically, the following four groups of determination processes are executed, and when all the conditions in each group are satisfied, it is determined that the state is a corresponding state, and a corresponding 2-bit value is output.

If E _C1 > E _C2 and E _C1 > E _C4 and E _C3 > E _C2 and E _C3 > E _C4 , it is determined that the state is “1” and a value 1 is output.

If E _C2 > E _C1 and E _C2 > E _C3 and E _C4 > E _C1 and E _C4 > E _C3 , it is determined that the state is “2”, and a value 2 is output.

If E _C1 > E _C2 and E _C1 > E _C3 and E _C4 > E _C2 and E _C4 > E _C3 , it is determined that “state 3”, and value 3 is output.

  If none of the above applies, it is determined that the state is “state 4”, and a value of 4 is output.

  And the code | symbol determination parameter calculation means 140 sets either the value 1-the value 4 according to the said determination result for each acoustic frame unit (S403).

  If any of the values 1 to 4 is set as a result of the determination, the phase determination table S (p) is further updated according to the following [Equation 24] (S404).

[Formula 24]
When it is determined that the state is “1” and the value 1 is output, S (p) ← S (p) + E _C1 + E _C3
When it is determined that the state is “2” and the value 2 is output, S (p) ← S (p) + E _C2 + E _C4
When it is determined that the state is “3” and the value 3 is output, S (p) ← S (p) + E _C1 + E _C4
When it is determined that the state is “state 4” and the value 4 is output, S (p) ← S (p) + E _C2 + E _C3

In FIG. 9, the code determination parameter calculation unit 140 stores a candidate for the optimum phase in the candidate code table (S304). Specifically, the value of the phase number p that maximizes the value of S (p) recorded in the phase determination table, the code of any of the four values determined in S407 and S408, and the sound frame The values of E _{C1 to} E _C4 corresponding to the components of the frequency band to be changed, which are calculated by executing the processing according to the above [Equation 23], are stored in the candidate code table as optimum phase candidates.

  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 p recorded in the candidate storage table is the optimum phase, and the code recorded in the candidate storage table is output. (S306).

  In FIG. 8, when the parity check results in pass, the additional information extraction means 160 adds 1 bit to the previous 7 bits and outputs (S208). Here, if the inspection result is acceptable, there is a high possibility that the eighth bit used for collation was an error detection bit. Then, the previous 7 bits are considered to be 7 bits in the original additional information. Therefore, by adding bit 0 to the 7 bits from the head, it is output as one word in the ASCII code. On the other hand, if there is a mismatch, there is a high possibility that the eighth bit used for collation is not an error detection bit. Then, it is considered that the 7 bits held at that time are shifted from the 7 bits in the original additional information. In this case, the first 2 bits are discarded, and processing for obtaining new 2 bits obtained by the processing from S201 to S206 is performed.

  In the processing of S208, the additional information extraction unit 160 determines that the additional information extraction unit 160 determines 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 by 1 byte ( It is recognized in units of 8 bits), character information is recognized according to the ASCII code, and displayed on a screen of a display device (not shown).

(2.3. Method for reliably embedding information even if the signal component is small)
The above processing is executed regardless of the magnitude of the signal component of the frequency band to be changed, but if the original signal component does not exist or is too small, the signal component necessary for extraction is obtained. Therefore, information cannot be properly extracted, resulting in an extraction error. Therefore, a method for enabling signal embedding so that information can be appropriately extracted even when the frequency band component of the original signal is small will be described.

  In this case, the information embedding process in the embedding apparatus shown in FIG. 2 is also performed according to the flowchart of FIG.

  Therefore, as a process for setting the state 1 to the state 4 in S104, first, the fixed value V calculated according to the following [Equation 25] is set as the intensity of the component of the change target frequency band.

[Formula 25]
V = {0.5 · Lev · C / P} ^1/2

  In the case of state 1, after executing the process according to the above [Equation 19], the process according to the following [Equation 26] is executed.

[Formula 26]
For each component of j = m to m + P−1, Al ′ (1, j) = F (j−m) · Al (1, j) · V / {Al (1, j) ² + Bl (1, j ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · V / {Al (1, j) ² + B1 (1, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (1, j) = F (j−m−P) .Al (1, j) .V / {Al (1, j) ² + Bl (1 , J) ² } ^1/2
Bl ′ (1, j) = F (j−m−P) · Bl (1, j) · V / {Al (1, j) ² + Bl (1, j) ² } ^1/2

  In the case of the state 2, after executing the process according to the above [Equation 20], the process according to the following [Equation 27] is executed.

[Formula 27]
For each component of j = m to m + P−1, Al ′ (3, j) = F (j−m) · Al (3, j) · V / {Al (3, j) ² + Bl (3, j ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · V / {Al (3, j) ² + B1 (3, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (3, j) = F (j−m−P) · Al (3, j) · V / {Al (3, j) ² + Bl (3 , J) ² } ^1/2
Bl ′ (3, j) = F (j−m−P) · B1 (3, j) · V / {Al (3, j) ² + B1 (3, j) ² } ^1/2

  In the case of the state 3, after executing the process according to the above [Equation 21], the process according to the following [Equation 28] is executed.

[Formula 28]
For each component of j = m to m + P−1, Al ′ (1, j) = F (j−m) · Al (1, j) · V / {Al (1, j) ² + Bl (1, j ² } ^1/2
Bl ′ (1, j) = F (j−m) · B1 (1, j) · V / {Al (1, j) ² + B1 (1, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (3, j) = F (j−m−P) · Al (3, j) · V / {Al (3, j) ² + Bl (3 , J) ² } ^1/2
Bl ′ (3, j) = F (j−m−P) · B1 (3, j) · V / {Al (3, j) ² + B1 (3, j) ² } ^1/2

  In the case of the state 4, after executing the process according to the above [Equation 22], the process according to the following [Equation 29] is executed.

[Formula 29]
For each component of j = m to m + P−1, Al ′ (3, j) = F (j−m) · Al (3, j) · V / {Al (3, j) ² + Bl (3, j ² } ^1/2
Bl ′ (3, j) = F (j−m) · B1 (3, j) · V / {Al (3, j) ² + B1 (3, j) ² } ^1/2
For each component of j = m + P to M−1, Al ′ (1, j) = F (j−m−P) .Al (1, j) .V / {Al (1, j) ² + Bl (1 , J) ² } ^1/2
Bl ′ (1, j) = F (j−m−P) · Bl (1, j) · V / {Al (1, j) ² + Bl (1, j) ² } ^1/2

(3. When using a mobile phone as an extraction device)
In the above embodiment, the case where the number N ′ of samples of one acoustic frame in the extraction device is N / 2 has been described. As described above, when a portable game machine, a PDA, a PHS terminal, or the like is employed as the hardware of the extraction device, the sampling frequency is ½ of the sampling frequency of the original acoustic signal. This is because the number of samples needs to be halved to match the number of seconds. When a normal mobile phone is used as the extraction device, the recording function of the mobile phone has a sampling frequency of 8 kHz, and the number of samples of one acoustic frame needs to be proportional to 743.

  As described above, when the number of samples N ′ = 743, in the flowchart of FIG. 8, the reference frame acquisition unit 110 generates an acoustic frame composed of 743 samples from the acoustic signal input from the acoustic signal input unit 100. Extract as a reference frame (S201). Therefore, the sound frame holding means 170 always stores two reference frames (continuous 1486 samples).

  Similar to the above embodiment, the acoustic frame processed by the embedding device is divided into a reference frame that is set adjacently without interruption from the beginning, and a phase change frame in which the phase has been changed.

  In S401 of FIG. 10, the frequency conversion unit 130 performs frequency conversion on each read sound frame to obtain a spectrum. At this time, N is replaced with 1024 in [Formula 5]. At this time, the samples Xl (i) and Xr (i) in [Formula 5] are all dummy up to i = 743 to 1023, and “0” is used as the value thereof. Note that W (1, i) to W (3, i) used in [Formula 5] are obtained by replacing N in [Formula 1] to [Formula 3] with 743, respectively. In [Formula 5], N is set to 1024 instead of 743 because a fast equivalent Fourier transform algorithm (FFT) is used to execute a calculation equivalent to [Formula 5]. This is because the speed is increased by the constraint that N is a power of 2. Therefore, when the discrete Fourier transform of [Formula 5] is executed according to the calculation formula, N can be set to 743 as it is, but it requires a calculation load of slightly more than two digits as compared with the FFT. Thus, it is not suitable for applications that are executed by a portable terminal with limited CPU performance. In addition, since FFT is an essential signal processing for decoding and playing back various video and audio data on a mobile terminal, there are not a few mobile terminals equipped with dedicated hardware. Even if it is low, there is a possibility that it can be executed at the same speed as a general-purpose PC by dedicated hardware.

In S402, the intensity value E _C1 and the intensity value E _C2 are used to determine what state the component of the frequency band to be changed is, but the intensity value E _C1 according to [Equation 13]. When calculating the intensity value E _C2 , m = 220 and M = 441 are executed. Further, when calculating the intensity values E _{C1 to} E _C4 according to [Equation 23] in the second embodiment, m = 220 and M = 441 are executed. This is not due to the number of samples 743 in one acoustic frame, but due to a shift caused by performing Fourier transform using 1024 samples, j = 220 (= 160 × 1024/743) corresponds to 1.7 kHz. This is because = 441 (= 320 × 1024/743) corresponds to 3.4 kHz. In other cases, the processing is performed in the same manner as in the case of N ′ = N / 2.

  Here, FIG. 16 shows the concept of the frequency domain at the time of embedding, at the time of extraction corresponding to the sampling frequency 22.05 kHz, and at the time of extraction corresponding to the sampling frequency 8 kHz. In FIG. 16, the vertical axis represents frequency. The number shown on the right side of each block is the value of j proportional to the frequency.

  At the time of embedding, since 4096 samples are subjected to Fourier transform, 2048 frequency component values are obtained, which correspond to 0 Hz to 22.05 kHz. Among these, data is embedded between j = 160 to 320 corresponding to 1.7 kHz to 3.4 kHz, and the other is the original sound.

  At the time of extraction corresponding to a sampling frequency of 22.05 kHz, since 2048 samples are subjected to Fourier transformation, 1024 frequency component values are obtained, which correspond to 0 Hz to 11.025 kHz. Among these, the value of j corresponding to 1.7 kHz to 3.4 kHz results in data extraction from j = 160 to 320 as in the case of embedding and sampling frequency of 22.05 kHz. , Others are ignored.

  At the time of extraction corresponding to the sampling frequency of 8 kHz, one acoustic frame is 743 samples. As described above, since a dummy sample is added and Fourier transform is performed on 1024 samples, 512 frequency components are included. A value is obtained. Since the sampling frequency is 8 kHz, these values correspond to 0 Hz to 4 kHz. For this reason, the value of j corresponding to 1.7 kHz to 3.4 kHz is j = 220 to 441, unlike the extraction at the time of embedding and the sampling frequency of 22.05 kHz. Data is extracted from j = 220 to 441, and others are ignored.

  Further, FIG. 17 shows signal waveforms at the time of embedding and extraction. In FIG. 17, the horizontal axis represents the time axis, and the vertical axis represents the amplitude intensity. At the time of embedding, an embedding process is performed on the acoustic signal sampled at 44.1 kHz as shown in FIG. 17A by the embedding device shown in FIG. 2, and the acoustic as shown in FIG. A signal is obtained. Then, sampling is performed at 22.05 kHz or 8 kHz by the extraction device shown in FIG. 7, and an acoustic signal as shown in FIG. 17C is obtained. Then, extraction processing is performed by setting one acoustic frame to have the same time as the acoustic frame at the time of embedding.

(4. Other)
As mentioned above, although it limited about the suitable embodiment of the present invention, the present invention is not limited to the above-mentioned embodiment, and various modifications are possible. For example, in the above embodiment, the number of samples of one acoustic frame is N = 4096, but N = 2048, 1024, 512, etc. may be set. As a result, the number of sound frames per same time is doubled, quadrupled, and quadrupled, and 2 to 8 times of information can be embedded as a whole.

  In the above embodiment, the order of adjacent odd-numbered bits and even-numbered bits in each bit of the latter half bit string is changed. However, this is not always necessary, and if the arrangement with the extraction side is made. The order may be changed according to some other rule.

It is a figure which shows the state of a change of the component of the change object frequency band in 1st Embodiment. It is a functional block diagram of an information embedding device for an acoustic signal. It is a figure which shows the time direction window function used by this invention. It is a figure which shows the frequency direction window function F (j) used by this invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a figure which shows the mode of preparation of a bit arrangement | sequence. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a 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 outline | summary in the setting which performs the error correction of 1 bit of the apparatus shown in FIG. It is a figure for demonstrating the conformity and parity check of a 16-bit arrangement | sequence equivalent to 1 word of additional information. It is a figure for demonstrating the conformity and parity check of a 16-bit arrangement | sequence equivalent to 1 word of additional information. It is a figure for demonstrating the conformity and parity check of a 16-bit arrangement | sequence equivalent to 1 word of additional information. It is a figure which shows the state of the change of the frequency component in 2nd Embodiment. It is a figure which shows the concept of the frequency area | region at the time of embedding, the extraction of sampling frequency 22.05kHz, and the extraction of sampling frequency 8kHz. It is a figure which shows the mode of the signal waveform at the time of embedding and extraction.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Acoustic frame reading means 20 ... Frequency conversion means 30 ... Frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified acoustic frame output means 60 ... Storage means 61 ... Acoustic signal storage unit 62 ... additional information storage unit 63 ... modified acoustic signal storage unit 70 ... bit array creation means 100 ... acoustic signal input means 110 ... reference frame acquisition means 120 ... phase Change frame setting means 130 ... frequency conversion means 140 ... code determination parameter calculation means 150 ... code output means 160 ... additional information extraction means 170 ... sound frame holding means

Claims (6)

Additional information is obtained from an analog acoustic signal reproduced by embedding additional information for each acoustic frame composed of a predetermined number N of samples in an inaudible state in advance for an acoustic signal digitized at a predetermined sampling frequency Fs. A device for extracting
Acoustic frame acquisition means for digitizing a predetermined section of the analog acoustic signal at a frequency Fs ′ smaller than the sampling frequency Fs to acquire an acoustic frame composed of a predetermined number N ′ of samples smaller than the predetermined number N;
Frequency conversion means for performing a frequency conversion on the acoustic frame using a window function to obtain a spectrum; and
Encoding that extracts a plurality of spectrum sets in a predetermined frequency range from the generated spectrum, calculates the spectrum intensity of each spectrum set, and extracts one or more embedded bit strings based on each spectrum intensity Means,
Additional information extracting means for extracting the additional information by converting the output bit string according to a predetermined rule in units of words;
An apparatus for extracting information from an acoustic signal, comprising: In claim 1,
The apparatus for extracting information from an acoustic signal, wherein the predetermined number N ′ satisfies a relationship of N ′ = N × Fs ′ / Fs. In claim 1 or claim 2,
The frequency converting means performs frequency conversion on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a third window function which are spectra corresponding to the first window function. A third window spectrum that is a spectrum corresponding to
The encoding means extracts a spectrum set of one or more predetermined frequency ranges from the generated first window spectrum, and extracts a spectrum set of one or more predetermined frequency ranges from the third window spectrum. The apparatus for extracting information from an acoustic signal is characterized in that one or more embedded bit strings are extracted based on each spectrum intensity. In any one of Claims 1-3,
The acoustic frame acquisition means changes a phase by moving a reference frame and a predetermined sample from a reference frame acquisition means for acquiring an acoustic frame composed of a predetermined number of samples as a reference frame from the acoustic signal. It is constituted by phase change frame setting means for setting a plurality of set sound frames as phase change frames,
The encoding means is based on the code determination parameter calculation means for calculating a code determination parameter based on the extracted spectrum set, and on the code determination parameter calculated in a past in-phase acoustic frame having a different reference frame, It is determined that one of the reference frame and the plurality of phase change frames has an optimal phase, and a predetermined code is determined based on the code determination parameter determined for the acoustic frame having the optimal phase. An apparatus for extracting information from an acoustic signal, characterized by comprising code output means for outputting. In claim 4,
The encoding means holds the optimum phase determined in the past, and when the optimum phase continues for a predetermined number of times or more, sets the phase as a definite phase,
When the definite phase exists, the acoustic frame acquisition unit extracts only one of the reference frame or the phase change frame corresponding to the deterministic phase, and the frequency conversion unit From the acoustic signal, wherein the encoding means outputs a predetermined code only from the extracted acoustic frame processed by the frequency conversion means. Information extraction device.   A program for causing a computer to function as an apparatus for extracting information from an acoustic signal according to any one of claims 1 to 5.

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