JP2008129198A - Information embedding device for acoustic signal and information extracting device from acoustic signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an information embedding device to an acoustic signal and an information extracting device from the acoustic signal, capable of expanding a receivable distance area by using a frequency band which suppresses degradation of reproducing quality, and which spreads further in distance than before. <P>SOLUTION: The acoustic signal is read by an acoustic frame unit of a predetermined size, and frequency conversion is performed by using a plurality of time direction windowing functions for each acoustic frame, and according to binary values of a bit sequence to be embedded, a state of a predetermined frequency band (1.7 to 3.4 kHz) is changed to " State 1" in which a front component strength is large, or "State 2" in which a rear component strength is large. At this time, in order to make the component strength large, the windowing function F which changes according to a value of a frequency direction so that the vale near a center of the predetermined frequency band may becomes large, is applied. <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.

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

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

In the methods described in Patent Documents 1 to 6, the amount of information that can be embedded is small, the sound quality is deteriorated to some extent, watermark information is lost by various signal processing, and watermark detection is difficult for analog copies. There is a problem. Therefore, the applicant uses the characteristic that the human sound source localization is dull in the low sound range, and changes the ratio of the two low frequency band components in the low frequency region of the acoustic signal according to the bit value of the attribute information. Thus, a method has been proposed in which attribute information (additional information) can be embedded in both monaural and stereo sound signals (see Patent Documents 7 and 8).
JP 2002-259421 A Japanese Patent Laid-Open No. 2003-157087 JP-A-11-145840 JP-A-11-219172 Japanese Patent No. 3321767 JP 2003-99077 A JP 2006-243340 A Japanese Patent Application No. 2005-147743

  However, the methods described in Patent Documents 7 and 8 have the following problems because information is embedded by changing the low frequency region. The first problem is that, in the case of monaural reproduction, the sound on the other channel side to be canceled is not generated, so that the quality is unbearable for listening. The second problem is that the low-frequency portion has a low speaker reproduction capability, and the low-frequency sound wave has a large attenuation in spatial propagation. Third, in the case of low frequencies, the lower limit cannot be lowered due to restrictions on speaker playback capability and microphone sensitivity in the low sound direction, and the sound source movement is noticeable in stereo playback for the high sound direction. Therefore, the upper limit cannot be raised, the frequency range that can be used for information embedding is limited, and the information embedding efficiency is poor.

  However, it is conceivable that a frequency range that is equal to or higher than a predetermined frequency and in which the human auditory sensitivity slightly falls is set as the embedded range. In this case, the frequency range below the embedded range is a portion where the human auditory sensitivity is high. Therefore, it is desirable to maintain the original sound. However, when the original sound is maintained, the frequency component also affects the embedding area, so that there is a problem that the embedded information cannot be extracted correctly.

  Accordingly, the present invention reduces the degradation of reproduction quality and uses a frequency band that can be propagated farther than before, thereby expanding the receivable distance range, and an information embedding device for an acoustic signal, An object is to provide an apparatus for extracting information from an acoustic signal.

  In order to solve the above-described problem, the present invention is an apparatus for embedding additional information in an inaudible state with respect to an acoustic signal composed of a time-series sample sequence, and a predetermined number of samples are embedded from the acoustic signal. A spectrum corresponding to the first window function by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, and an acoustic frame reading means for reading as an acoustic frame. Frequency conversion means for generating a first window spectrum, a second window spectrum that is a spectrum corresponding to the second window function, a third window spectrum that is a spectrum corresponding to the third window function, and each of the generated windows A spectrum set corresponding to a predetermined frequency band having an upper limit or a lower limit of the telephone line band upper limit is extracted from the spectrum, and the information array of the additional information to be embedded is extracted. Based on the above, among the extracted three sets of spectrum sets, the ratio of the spectral intensities of the first window function at the front and the third window function at the rear is set so that the value near the center in the frequency range of the spectrum set becomes large. A frequency function changing means for applying and changing a window function F that changes in accordance with a value in the frequency direction, and deleting a component of the second window function at the center, and each window spectrum including the changed spectrum intensity A device for embedding information with respect to an acoustic signal is provided that includes frequency inverse transform means for performing frequency inverse transform to generate a modified acoustic frame and modified acoustic frame output means for sequentially outputting the generated modified acoustic frames. .

  According to the present invention, there is provided an apparatus for extracting additional information embedded in an inaudible state in advance from an acoustic signal, wherein a predetermined section of the acoustic signal is digitized and an acoustic composed of a predetermined number of samples. An acoustic frame acquisition means for acquiring a frame; a first window spectrum that is a spectrum corresponding to the first window function by performing frequency conversion on the acoustic frame using a first window function and a third window function, respectively; Frequency conversion means for generating a third window spectrum, which is a spectrum corresponding to the third window function, and a spectrum corresponding to a predetermined frequency band whose upper limit or lower limit is a telephone line band upper limit from the generated window spectra. A set is extracted, and the sum of the spectral intensities for each spectrum set is calculated as a value near the center in the frequency range of the spectrum set. An encoding means for applying a window function F that changes in accordance with a value in the frequency direction so as to be determined, and outputting a predetermined code based on a ratio between the spectrum sets of the total value, and the output There is provided an apparatus for extracting information from an acoustic signal having additional information extraction means for extracting additional information by converting an information array corresponding to the generated code according to a predetermined rule.

  According to the present invention, the acoustic signal is divided into predetermined sections, and based on the value of the information array of the additional information to be embedded, the head portion of the predetermined frequency band having the upper limit or the lower limit of the telephone line band upper limit in the predetermined section Since the ratio of the spectral intensity of the rear part is changed by applying the window function F that changes according to the value in the frequency direction so that the value near the center in the frequency range of the spectrum set is increased, the monaural In addition to reducing the degradation in reproduction quality during reproduction, it is possible to expand the receivable distance range.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Basic concept of the present invention)
First, the basic concept of the present invention will be described. As described above, in the patent documents 7 and 8, the present applicant changes the ratio of two low-frequency band components in the low-frequency region of the acoustic signal according to the bit value of the attribute information, so that the monaural, stereo In any of these acoustic signals, a method capable of embedding additional information is proposed. In particular, Patent Document 8 proposes a method of embedding information by dividing it in the time direction. As described above, the present invention is similar to Patent Document 8 in that the strength of the front and rear portions in a predetermined section is changed for a predetermined frequency component in accordance with information to be embedded.

  However, in the method shown in Patent Document 8, since the low frequency component of about 200 to 300 Hz or less is used as the predetermined frequency component, the problem described as the above problem occurs.

  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.

(2. Information embedding device for acoustic signals)
FIG. 1 is a functional block diagram showing the configuration of an information embedding device for an acoustic signal according to the present invention. In FIG. 1, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a 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 an additional information reading means. The apparatus shown in FIG. 1 can deal with both a stereo sound signal and a monaural sound signal. Here, a case where processing is performed on a stereo sound signal will be described.

  The sound frame reading means 10 has a function of reading a predetermined number of samples as one frame from each channel of the original stereo sound signal to be embedded with additional information. The frequency conversion means 20 has a function of generating a frame spectrum by frequency-converting the frame of the acoustic signal read by the acoustic frame reading means 10 by Fourier transformation or the like. The frequency component changing means 30 extracts three sets of spectrum sets corresponding to a predetermined frequency band whose upper limit or lower limit is the telephone line band upper limit from the generated frame spectrum, and based on the additional information extracted from the additional information storage unit 62 Thus, it has a function of changing the intensity of each spectrum set. The frequency inverse transform means 40 has a function of generating a modified acoustic frame by performing frequency inverse transform on a frame spectrum including a spectrum set whose intensity has been changed. The modified sound frame output means 50 has a function of sequentially outputting the generated modified sound frames. The storage means 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded with additional information, an additional information storage unit 62 that is configured as a bit array and stores additional information embedded in the stereo acoustic signal, and an additional information It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after information is embedded, and stores various information necessary for other processing. The additional information reading means 70 has a function of extracting additional information from the additional information storage unit 62 and adding one bit of error detection bits to the extracted additional information to create an 8-bit bit array. The additional information is information that should be added to the sound information and embedded, and includes attribute information such as a title and artist name, and other information other than the attribute information. Each component shown in FIG. 1 is actually realized by installing a dedicated program in hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

(3. Processing operation of the embedding device)
Next, the processing operation in the first embodiment of the information embedding device for the acoustic signal shown in FIG. 1 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. In order to obtain better effects as a plurality of window functions, in the present embodiment, the first window function W (1, i) and the second window function W (2) as shown in FIGS. , I) and a third window function W (3, i) are prepared so that the extraction side can easily recognize them. 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. 2 (a) 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. 2 (c), 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. 2A has a signal component remaining at 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. 2 (d), the front part has a minimum value of 0 and a predetermined value in the rear part. The maximum value 1 is set at the position of 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. 2 (a) is removed from the front signal component as shown in FIG. 2 (h). The signal component remains and becomes a Fourier transform target. Thus, after extracting the front part, the central part, and the rear part, the Fourier transform is executed, so that spectra corresponding to the front part, the central part, and the rear part are obtained. In order to embed a bit value in one acoustic frame, the bit value is originally divided into two parts, a front part and a rear part. However, on the extraction side, it is not always possible to read a signal synchronously. Therefore, in order to clearly distinguish the front part from the rear part, in the present invention, the signal component in the central part is always removed at the time of embedding, and the front part and the rear part are separated in time (however, at the time of extraction) You only need to analyze the front and rear, and ignore the middle). In the window function used in the present invention, 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 window function W (4, i) as shown in FIG. 2 (e) is used.

  In the present embodiment, the sound frame is read in duplicate. That is, a predetermined number of samples are redundantly read in the odd-numbered sound frames and the even-numbered sound frames. As 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. 2, the horizontal axis is the time axis (i). As described later, i is a serial number assigned to N samples in each acoustic frame, and is proportional to time t. 2A, 2F, 2G, 2H, and 2I, the vertical axis indicates the amplitude value (level) of the signal. 2B to 2E, 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. 2 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 spectrum of the signal component obtained by applying a window function to the signal component of each acoustic frame is obtained. Subsequently, the frequency component changing unit 30 extracts three spectrum sets from a predetermined frequency region of the generated spectrum. As described above, in this embodiment, the “telephone high frequency band” of “1.7 kHz to 3.4 kHz” is extracted.

  Here, FIG. 3 shows the state of the frequency component according to the present embodiment in units of one acoustic frame. In each acoustic frame shown in FIG. 3, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction. The shaded portion indicates a portion where the frequency component exists (greater than 0), and the darker the shaded portion, the stronger the component strength. In the present embodiment, the codes to be embedded are distinguished based on the distribution of the front and rear components in the telephone high frequency band.

  In the present embodiment, three states are set by changing the state of the component in the telephone high frequency band of each acoustic frame with respect to the frequency component in the original state as shown in FIG. Specifically, both the front and rear components are removed (hereinafter referred to as “state 0”), the front strength is high (hereinafter referred to as “state 1”), and the rear strength is high ( Hereinafter, the state is changed to any one of “state 2”. Among these, since state 0 is used when a writing error occurs, state 1 and state 2 enable binary (1 bit) information to be embedded in one acoustic frame.

  When the state is set to 0, all the components of the telephone high frequency band of the L-ch signal are removed, and a component equivalent to the removed component is added to the telephone high frequency band of the R-ch signal, and the result is shown in FIG. Change to the state shown. When the state 1 is set, the intensity of each spectrum set in the front of the telephone high frequency band of the L-ch signal is increased, and the intensity of each spectrum set in the front of the telephone high frequency band of the R-ch signal is decreased. . In addition, the components of each spectrum set at the center and the rear of the telephone high frequency band of the L-ch signal are removed, and a component equivalent to the removed component is added to the telephone high frequency band of the R-ch signal. In this way, the state is changed to the state shown in FIG.

  When setting to state 2, the component of each spectrum set in the front and center of the telephone high frequency band of the L-ch signal is removed, and a component equivalent to the removed component is changed to the telephone high frequency band of the R-ch signal. to add. Further, the intensity of the spectrum set at the rear of the telephone high frequency band of the L-ch signal is increased, and the intensity of the spectrum set at the rear of the telephone high frequency band of the R-ch signal is weakened. In this way, the state is changed to the state shown in FIG.

The frequency component changing unit 30 performs a process of changing the intensity of the spectrum set according to the bit value read by the additional information reading unit 70 for the A type acoustic frame. In the present embodiment, additional information is read bit by bit, and one of two values is embedded in one acoustic frame. The frequency component changing means 30 changes the state of the telephone high frequency band component to either state 1 or state 2 according to the read value 1 and value 2. Specifically, first, the frequency component changing means 30 performs real part Al (1, j), Al (3, j), imaginary part Bl (1, j) of the left channel obtained by the above [Equation 5], By using Bl (3, j), real part Ar (1, j), Ar (3, j), imaginary part Br (1, j), Br (3, j) of the right channel, 7], the intensity values E ₁ and E ₂ are calculated.

[Formula 7]
E ₁ = Σ _{j = m, M-1} {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } · F (j−m) · C
E ₂ = Σ _{j = m, M−1} {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } · F (j−m) · C

  F (j) in the above [Equation 7] is a window function that changes according to the value in the frequency direction, and is defined by the following [Equation 8].

[Formula 8]
F (j) = 1.0− (j−Pr) ² / Pr ²

  In the above [Formula 8], Pr = P / 2. Note that P = M−m. 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 telephone high 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 telephone high frequency band.

  Further, C in [Expression 7] is a fixed coefficient, and corrects the strength to be reduced by using the frequency window function. The fixed coefficient C is defined by the following [Equation 9].

[Formula 9]
C = 1.0 / Σ _{j = 0, P-1} F (j)

In the above [Expression 7], m is the number of the lower limit component of the telephone high frequency band, and M is the number of the component immediately above the upper limit of the telephone high frequency band. In this embodiment, in order to set the telephone high frequency band to “1.7 kHz to 3.4 kHz”, m = 160 and M = 320 are set. Therefore, in this case, P = 160. E ₁ and E ₂ calculated by the above [Equation 7] indicate the intensity values of the front and rear spectrum sets in the telephone high frequency band of the acoustic frame. Subsequently, it is determined whether or not the intensity values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev. The level lower limit value Lev is set to 0.25 when the maximum amplitude value of the acoustic signals Xl (i) and Xr (i) is normalized to 1 and M = 320. This value of Lev = 0.25 is a level at which the resistance to analog conversion can be maintained empirically, and if the component intensity is small, it will be reduced as appropriate, but in that case, the detection accuracy also decreases due to analog conversion. It will be.

The reason why it is determined whether the intensity values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev is that if the signal intensity is small, even if the signal is changed, the change cannot be detected on the extraction side. is there. That is, it is determined whether or not each spectrum component in the telephone high frequency band can be made sufficiently large.

When E ₁ is larger than the level lower limit value Lev and the value to be embedded is “value 1”, the state according to the following [Equation 10] is executed to change the state of the telephone high frequency band to “state 1”, that is, Then, the state is changed as shown in FIG.

[Formula 10]
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
Furthermore, for each component of j = m 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) · 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)) · Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Br ′ (1, j) = (1.0−F (j−m)) · B1 (1, j) · E (1, j) / {Al (1, j) ² + B1 (1, j) ² } ^1/2

  In the above [Expression 10], both of Al ′ (3, j) and Bl ′ (3, j) are set to 0 in j = m to M−1. As shown in the upper part of FIG. 3C, this indicates that each component at the rear part of the telephone high frequency band is set to 0 in L-ch. Since it is sufficient if the difference from the part can be clarified, it is not necessarily required to be 0, and a sufficiently small value may be used.

When E ₂ is larger than the level lower limit value Lev and the value to be embedded is “value 2”, the process according to the following [Equation 11] is executed to change the state of the telephone high frequency band to “state 2”, that is, Then, the state is changed to the state shown in FIG.

[Formula 11]
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
Furthermore, for each component of j = m 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) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = F (j−m) · Bl (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Ar ′ (3, j) = (1.0−F (j−m)) · Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Br ′ (3, j) = (1.0−F (j−m)) · B1 (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2

  In the above [Formula 11], both of Al ′ (1, j) and Bl ′ (1, j) are set to 0 in j = m to M−1. As shown in the upper part of FIG. 3D, this indicates that each component at the front part of the telephone high frequency band is set to 0 in L-ch, but “state 2” is the rear part of the telephone high frequency band. Since it is sufficient if the difference between the two can be clarified, it is not always necessary to set the value to 0, and may be a sufficiently small value.

  When embedding a value, if the component strength of the telephone high frequency band on the embedding side is too small, even if the signal is changed, the change cannot be detected on the extraction side. Therefore, in such a case, in order to clarify the difference from the acoustic frame in which the information is embedded, processing according to the following [Equation 12] is executed to remove the component of the telephone high frequency band. This is a “state 0” state as shown in FIG.

[Formula 12]
For each component of j = 0 to M−1, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
Al ′ (2, j) = 0
Bl ′ (2, j) = 0
Al ′ (3, j) = 0
Bl ′ (3, 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
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
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

  As described above, since it is necessary to change the telephone high frequency band in accordance with the bit value to be embedded for the A type sound frame, the frequency component changing means 30 performs the above [Formula 10] [Formula 11]. The process according to [Formula 12] is executed. However, since the B type acoustic frame is used to prevent discontinuity at both ends when only the A type acoustic frame is used, the telephone high frequency band component is changed according to the bit value. There is no need. Therefore, the frequency component changing unit 30 executes processing according to the following [Equation 13] for the B type sound frame, and always removes the telephone high frequency band component of the left signal.

[Formula 13]
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 reverse conversion means 40 performs the process of performing frequency reverse conversion on the frame spectrum in which the state of the telephone high frequency band component is changed to obtain a modified acoustic frame. 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 acoustic frame, the frequency inverse transform means 40 performs the real part Al ′ (1) of the left channel of the spectrum obtained by the above [Formula 10] [Formula 11] [Formula 12]. ), J), imaginary part Bl ′ (1, j), etc., real part Ar ′ (1, j), etc. of the right channel, imaginary part Br ′ (1, j), etc. The processing according to the above is performed to calculate Xl ′ (i) and Xr ′ (i). For frequency components that are not modified in the above [Equation 10], [Equation 11], and [Equation 12], the original frequency component Al (1, j) or the like is used as Al ′ (1, j) or the like. .

[Formula 14]
Xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) -Σ j Bl' (1, j) · sin (2πij / N)} + 1 / N · { Σ _j Al ′ (2, j) · cos (2πij / N) −Σ _j Bl ′ (2, j) · sin (2πij / N)} + 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 14], Σ _{j = 0,} ... _{, N−1} is shown as Σ _{j in} order to prevent the expression from becoming complicated. The terms “+ Xlp (i + N / 2)” in the first equation and “+ Xrp (i + N / 2)” in the second equation in the [Equation 14] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. According to the above [Equation 14], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the type A modified acoustic frame are obtained.

  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 13]. , Using the real part Ar ′ (4, j) and the imaginary part Br ′ (4, j) of the right channel, the processing according to the following [Equation 15] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For frequency components that are not modified in the above [Equation 13], in the following [Equation 15], 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 15]
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 15], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified acoustic frame are obtained.

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

  Next, the overall flow of processing of the information embedding device for the acoustic signal shown in FIG. 1 will be described with reference to the flowchart of FIG. Each component which comprises the apparatus shown in FIG. 1 cooperates, and performs the process according to FIG. 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 telephone high frequency band component is removed by the frequency component changing unit 30, the frequency is inversely converted by the frequency inverse converting unit 40, and then output by the modified acoustic frame output unit 50.

  In FIG. 5, first, the additional information reading means 70 adds one bit of error detection bits to the additional information extracted from the additional information storage unit 62 to create an 8-bit bit array, and uses this as one word. Set (S101). Specifically, one word is read into a register in a computer used as an information embedding device for an acoustic signal.

  Next, the frequency component changing unit 30 performs a process of reading one bit from one word 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 telephone high frequency band 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 calculates E _{1 to} E ₂ by the above [Equation 7]. Then, the processing according to the above [Formula 10] and [Formula 11] is executed according to the value 1 and the value 2 which are the bit values of each bit received from the additional information reading means 70, and the state of the telephone high frequency band component is changed to “ Change to either state 1 "or" state 2 ".

If it is not possible to change to either “state 1” or “state 2” in S104, processing for setting the telephone high frequency band component to “state 0” is performed (S105). Specifically, the process according to the above [Equation 12] is executed. The case where the state cannot be changed to either “state 1” or “state 2” means that _{one of} the intensity values E ₁ and E ₂ corresponding to the value to be embedded is equal to or lower than the level lower limit value Lev. It is. For example, when the value 1 is embedded, E ₁ needs to be larger than the level lower limit value Lev. Therefore, if E ₁ is less than or equal to Lev, the process proceeds to S105.

  By executing the processing according to the above [Equation 12], the telephone high frequency band component of the frame spectrum of the left channel is the same as “0” in both the front spectrum set and the rear spectrum set. In the above [Equation 12], Al ′ (j) = Bl ′ (j) = 0 is set for each component of j = 0 to M−1, but “state 1”, “ The purpose is to make it possible to recognize that the state is not any of the states 2 ″. Therefore, if the value is sufficiently small, it is not necessarily set to 0. In addition, the values of the front and rear spectral components do not necessarily have to be the same, and it is sufficient that the difference is small.

  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 intensity ratio between the spectrum sets is changed by the processes of S104 and S105 (S106). 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 imaginary part Bl ′ (1, j), such as the real part Al ′ (1, j) of the left channel of the spectrum obtained by any one of the above [Formula 10], [Formula 11] and [Formula 12]. And the like, using the real part Ar ′ (1, j) of the right channel, the imaginary part Br ′ (1, j), etc., the processing according to the above [Equation 14] is performed, and Xl ′ (i), Xr ′ (I) is calculated.

  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 finished in this way, when the state is set to 0, the acoustic frame reading means 10 reads the next acoustic frame (S103). On the other hand, when either state 1 or state 2 is set, the frequency component changing unit 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 process corresponding to each bit of the data of one word read in S101 is completed, the process returns from S102 to S101, and the process of creating the bit array by reading the next word of the additional information is performed. When the processing is completed for all the words of the additional information, the processing returns to the first word of the additional information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

  In this embodiment, the case has been described where the additional information is 7 bits per word, and 1 bit for error detection is added and processed as 8 bits per word. However, as long as there is an agreement with the extraction side, the additional information One word can be recorded in an arbitrary number of bits.

  Of the left channel of the modified acoustic signal obtained as described above, in the portion where the additional information is embedded, the telephone high frequency band component has only three distributions of the state 0 to the state 2 described above. . However, the components in the frequency band other than the telephone high frequency band remain the original acoustic signal, and thus have various distributions based on the setting of the producer. Further, as shown in the above example, when a stereo sound signal is used, the telephone high frequency band component changed in the left channel is obtained from the processing of [Expression 10], [Expression 11], and [Expression 12]. As is obvious, it is always added to the telephone high frequency band component of the right channel. Therefore, since the right channel supplements the deleted component in the left channel, there is no signal degradation when viewed as both channels as a whole.

(4. Device for extracting information from acoustic signals)
Next, an apparatus for extracting information from an acoustic signal according to the present invention will be described. FIG. 6 is a block diagram showing an embodiment of an apparatus for extracting information from an acoustic signal according to the present invention. In FIG. 6, 100 is an acoustic signal input unit, 110 is a reference frame acquisition unit, 120 is a phase change frame setting unit, 130 is a frequency conversion unit, 140 is a code determination parameter calculation unit, 150 is a code output unit, and 160 is additional information. Extraction means 170 is an acoustic frame holding means.

  The acoustic signal input unit 100 has a function of acquiring and inputting a flowing sound as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. As the microphone, any microphone can be used as long as it can detect a high-frequency band component of a telephone, and it can be monaural omnidirectional or stereo directional. 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 frequency components corresponding to the telephone high frequency band from the generated frame spectrum, and calculates intensity values E _C1 and E _C2 corresponding to each spectrum set based on the following [Equation 16]. The intensity values E _C1 and E _C2 are used as code determination parameters, and a function of determining a predetermined state based on the ratio of the code determination parameters E _C1 and E _C2 is provided. The following [Formula 16] is obtained by deleting the right channel component in the above [Formula 7], and does not refer to the right channel component at the time of extraction.

[Formula 16]
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 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 ternary 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. 6 is actually realized by mounting a dedicated program on hardware such as a small computer having an information processing function and its peripheral devices. In particular, in order to achieve the object of the present invention more easily, it is desirable to use a portable terminal device as hardware.

(5. Processing operation of extraction device)
Next, the processing operation of the apparatus for extracting information from the acoustic signal shown in FIG. 6 will be described with reference to the flowchart of FIG. First, in this apparatus, the average code levels HL1 and HL2, the phase determination table S (p), and the non-code counter Nn are initialized (S200). These will be described. The average code levels HL1 and HL2 are telephone high-frequency band components calculated by the above [Equation 16] for an acoustic frame (hereinafter referred to as an effective frame) determined to be embedded with values 1 and 2. Intensity values E _C1 and E _C2 are average values, that is, given by the average value of these values in the past effective frame, and the initial value is set to 0.1. The phase determination table S (p) is a table for determining the phase, and p takes an integer value of 0 to 5. The initial value is set to S (p) = 0. The non-sign counter Nn is a counter for the number of frames that are determined to have a low signal level and non-sign (information indicating an embedding error), and is set to Nn = 0 in the initial state.

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

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 100 as a reference frame (S201). Specifically, the reference frame is extracted and read into the acoustic frame holding unit 170. The number of samples of one sound frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the sound frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples as reference frames. The acoustic frame holding means 170 can store two reference frames as described above, and when a new reference frame is read, the old reference frame is discarded. Therefore, the sound frame holding means 170 always stores two reference frames (continuous 8192 samples).

  The acoustic frame processed by the embedding device can be divided into a reference frame that is set adjacently without interruption from the head, and a phase change frame in which the phase is changed. For the reference frame, after setting sample number 1 to sample number 4096 as the first reference frame, the next reference frame is sample number 4097 to sample number 8192, and the next reference frame is sample number 8193 to sample number 12288. It is set without interruption. Then, for each reference frame, five phase change frames moved by 1/6 frame (about 683 samples) are set. For example, for the first reference frame, five phase change frames configured by 4096 samples starting from sample numbers 683, 1366, 2049, 2732, and 3413 are set. Subsequently, the frequency conversion unit 130 and the code determination parameter calculation unit 140 determine embedded information from each read sound frame and output a corresponding code (S202). The format of the information to be output is a total ternary format of value 1, binary 2 and value 0 corresponding to the case where additional information is embedded.

  Details of the code determination process in step S202 will be described with reference to the flowchart of FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain a frame spectrum (S301). This process is the same as the process in the frequency conversion means 20 shown in FIG. However, since only the left channel is used for extraction, the processing according to [Formula 5] is performed, and the imaginary part Bl (1) such as the real part Al (1, j) of the conversion data corresponding to the left channel is performed. , J) etc.

By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, the code determination parameter calculation unit 140 calculates the average code levels HL1 and HL2 (S302). Specifically, v1 that is an integrated value of the intensity values E _C1 for the acoustic frames that have been determined to be in the “state 1” in the past is represented by n1 that is the number of acoustic frames that have been determined to be in the past “state 1”. HL1 is calculated by dividing by 2, and v2 which is the integrated value of the intensity value E _C2 for the acoustic frame that has been determined to be in the past “state 2” is the acoustic frame that has been determined to be in the past “state 2”. HL2 is calculated by dividing by n2.

  Subsequently, the code determination parameter calculation unit 140 initializes the candidate code table (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 ternary code obtained from the states of the six acoustic frames.

Subsequently, the code determination parameter calculation means 140 uses the intensity value E _C1 and the intensity value E _C2 to determine what the state of the telephone high frequency band component is, that is, what value as a 1-bit value. A process is performed to determine whether or not embedded (S304). Specifically, as the predetermined values corresponding to the intensity values E _C1 and E _C2 , values obtained by multiplying the average code levels HL1 and HL2 by 0.01 are determined according to the following conditions, and the corresponding codes are set. Output.

If E _C1 > 0.01HL1 and E _C1 > 2 · E _C2 , the value 1 is output. If E _C2 > 0.01HL2 and E _C2 > 2 · E _C1 , the value 2 is output.

The code determination parameter calculation means 140 outputs either value 1 or value 2 according to the determination result for each acoustic frame (S307). If none of the above conditions is satisfied, it is determined that the state is “0”, and a value of 0 is output (S308). This “state 0” indicates an embedding error indicating that the recording of the values 1 and 2 has not been performed in the frame because the intensity values E ₁ and E ₂ are equal to or lower than the lower limit value Lev.

  If either value 1 or value 2 is output as a result of the determination, the phase determination table S (p) is further updated according to the following [Equation 17] (S309).

[Formula 17]
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

Subsequently, the code determination parameter calculation unit 140 saves the candidate for the optimum phase in the candidate code table (S310). Specifically, the value of the phase number p that maximizes the value of S (p) recorded in the phase determination table, one of the three values determined in S307 and S308, and the sound frame One of E _C1 and E _C2 corresponding to the telephone high frequency band component calculated by executing the processing according to the above [Equation 16] is stored in the candidate code table as a candidate for the optimum phase.

  Subsequently, it is determined whether or not the processing corresponding to all the phase numbers p has been completed (S311). 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 processed acoustic frame, and the process proceeds to S304. 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. (S312).

Returning to the flowchart of FIG. As a result of the process in S202, when a code corresponding to value 1 and value 2 is output, an update process of the parameter of the average code level is performed (S203). More specifically, the intensity values E _C1 and E _C2 are added to the integrated values v1 and v2 which are numerators when the average code levels HL1 and HL2 are calculated, respectively, and the integrated values v1 and v2 are updated. This is done by adding 1 to the numbers n1 and n2 to update the frame numbers n1 and n2. n1 and n2 are set so as not to exceed preset n _max (for example, 100).

  Subsequently, initialization processing of a non-sign counter is performed (S204). Specifically, the non-sign counter Nn = 0 is set in the same manner as in the initialization process of S200. Further, 1 bit corresponding to the output value is stored in the buffer (S205). Next, the bit counter is incremented by “1” (S206).

  Next, it is determined whether the bit counter is 7 or less or 8 or more. If the bit counter is 7 or less, the process returns to S201 to perform processing for extracting the next reference frame.

  If the bit counter is 8 or more, the last 8 bits are extracted from the bit string stored in the buffer, and parity calculation is performed with the previous 7 bits (S207). And it collates with the 8th bit among the last 8 bits extracted. As a result, if they do not match, the process returns to S201 to perform processing for extracting the next reference frame.

  If they match as a result of the collation, the additional information extraction means 160 adds 1 bit to the previous 7 bits and outputs it (S208). Here, if the collation results match, 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 1 bit is discarded, and processing for obtaining a new 1 bit is performed by the processing from S201 to S206.

  If parity check is performed in this way and the result passes, 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 the next parity check will be rejected, 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.

If the value 0 is output as a result of the process in S202, the non-sign counter is incremented (S210). Specifically, 1 is added to the value of the unsigned counter Nn. If the value of the non-sign counter Nn is equal to or greater than n _max , the process returns to S200 and initialization processing is performed. If the value of the non-code counter Nn is less than n _max , the process returns to S201 and the code extraction process is continued. By executing the processing shown in FIG. 7 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 processing, in order to extract additional information accurately in the extraction device, processing for correcting the phase and processing for correcting the lower limit threshold value for determining that the frame is an invalid frame are performed. Next, supplementary explanation will be given for these correction processes.

(6. About phase correction processing)
As described above, at the time of extraction, it is not always possible to read an acoustic signal corresponding to the acoustic frame embedded at the time of embedding. Therefore, the phase of the acoustic frame is shifted and read in a plurality of ways (six in this embodiment), the optimum phase is determined, and a code corresponding to the acoustic frame specified by the phase is output. Yes. For example, in the case of reading in six ways, the top acoustic frame is originally a sample of sample numbers 1 to 4096, but six pieces composed of 4096 samples starting from sample numbers 1, 683, 1366, 2049, 2732, and 3413 Are processed, and a code corresponding to the optimum acoustic frame is output. This phase correction process is performed centering on the processes in S303, S309, S310, S311, and S312.

(7. Lower limit correction process)
When the signal level is small, the magnitude of the intensity of the front part and the rear part cannot be determined, and misjudgment frequently occurs on the extraction side. For this reason, frames whose intensity values E _C1 and E _C2 are equal to or less than a predetermined threshold are determined to be invalid frames. The threshold at this time is used as an integrated value of frequency intensity for past effective frames. A correction process is performed using this. By varying the threshold value in this way, it is possible to accurately determine whether the frame is invalid or valid even if the signal level varies. This lower limit threshold correction process is performed centering on the processes in S302 and S203.

(8. In the case of monaural sound signal)
In the above embodiment, the case where the additional information is embedded in the left channel signal of the stereo sound signal having the left and right channels 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.

(9. Method that enables information to be embedded even if the signal component is small)
So far, in the processing described above E _1, E ₂ is required to be larger than a predetermined value, the E _1, E ₂ is the case of less than the predetermined value, it is impossible to embed information. Therefore, a method for enabling signal embedding even when E ₁ and E ₂ are not more than a predetermined value will be described below.

In this case, the information embedding process in the embedding apparatus shown in FIG. 1 is performed according to the flowchart of FIG. 5. However, since the intensity values E ₁ and E ₂ do not become lower than the level lower limit value Lev, the frequency component in S104 The changing means 30 does not determine the level. This is because, in this process, information is forcibly embedded even if the signal level is low, so it is not necessary to determine whether there is a portion with a low signal level where information cannot be embedded. .

  Therefore, as processing for setting the state 1 and the state 2 in S104, first, the fixed value V calculated according to the following [Equation 18] is set as the strength of the telephone high frequency band component.

[Formula 18]
V = {0.25 · Lev / (M−m)} ^1/2

  In the case of state 1, after executing the process according to the above [Equation 10], the process according to the following [Equation 19] is executed.

[Formula 19]
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 11], the process according to the following [Expression 20] is executed.

[Formula 20]
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

[Equation 19] When the values of Al ′ and Bl ′ calculated by [Equation 20] are used instead of Al and Bl in the above [Equation 7], E ₁ and E ₂ = Lev. It is not necessary to determine the magnitude relationship with Lev.

  As described above, even when information is embedded when the frequency component is small, the configuration of the extraction device for extracting information from the acoustic signal on the extraction side is the same as that in FIG. 6, and the processing operation is the flowchart in FIG. Is the same as

  In the above description, a case where processing is performed on a telephone high frequency band corresponding to “1.7 kHz to 3.4 kHz”, which is a predetermined frequency band whose upper limit is the upper limit of the telephone line band, has been described as an example. When using a device capable of generating and recording sound in a wider frequency range without using a fixed telephone or a mobile phone as an information extraction device, an information extraction device from an acoustic signal is realized using a telephone line. Processing may be performed on a super telephone frequency band corresponding to “3.4 kHz to 6.8 kHz”, which is a predetermined frequency band whose upper limit is the upper limit of the band. In this case, the values of m and M used in each equation including the above [Equation 7] are set to m = 320 and M = 640. Therefore, in this case, P = 320.

In all of the above, the case where the sampling frequency of the acoustic signal is 44.1 kHz and the length N of the acoustic frame is 4096 has been described, but the length N of the acoustic frame is 1/2 ^K (K is an integer of 1 or more). The amount of information that can be embedded per unit time (in the above description, 10 [bps]) can be expanded by 2 ^K times. At that time, if the sampling frequency of the acoustic signal is fixed to 44.1 kHz, the lower limit m and the upper limit M of the telephone high frequency band or super telephone frequency band must be set to be reduced to 1/2 ^K times (the same applies to P). 1/2 ^K times). Specifically, when N = 2048 is set, m = 80 and M = 160 in the case of the telephone high frequency band, and m = 160 and M = 320 in the case of the super telephone frequency band. [Bps]. When N = 1024, m = 40 and M = 80 in the case of the telephone high frequency band, and m = 80 and M = 160 in the super telephone frequency band, and the amount of information that can be embedded is 40 [bps]. . When N = 512 is set, m = 20 and M = 40 in the case of the telephone high frequency band, and m = 40 and M = 80 in the super telephone frequency band, and the amount of information that can be embedded is 80 [bps]. . Furthermore, it is theoretically possible to set N = 256, N = 128, etc. However, as a result of experiments, if N = 256 or less, the embedded area (time × frequency width) is too small and a calculation error occurs, so that stable extraction is possible. N = 512 was the setting limit.

It is a functional block diagram of an information embedding device for an acoustic signal according to the present invention. It is a figure which shows the time-axis direction window function used by this invention. It is a figure which shows the mode of the component change of 1 acoustic frame by the process of the apparatus shown in FIG. It is a figure which shows the frequency direction window function F used by this invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a flowchart which shows the detail of the code | symbol determination process of S202 of FIG.

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 ... additional information reading means 100 ... acoustic signal input means 110 ... reference frame acquisition means 120 ... phase Change frame setting means 130 ... frequency conversion means 140 ... code determination parameter calculation means 150 ... code output means 160 ... additional information extraction means 170 ... sound frame holding means

Claims (11)

A device for embedding additional information in an inaudible state with respect to an acoustic signal composed of a time-series sample sequence,
From the acoustic signal, acoustic frame reading means for reading a predetermined number of samples as an acoustic frame;
The first window function, the second window function, and the second window function, which are spectrums corresponding to the first window function, by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, respectively. Frequency conversion means for generating a second window spectrum that is a spectrum corresponding to, and a third window spectrum that is a spectrum corresponding to the third window function;
A spectrum set corresponding to a predetermined frequency band whose upper limit or lower limit is the telephone line band upper limit is extracted from each generated window spectrum, and the extracted 3 is extracted based on the value of the information array of the additional information to be embedded. Among the set of spectrum sets, the ratio of the spectrum intensity of the first window function at the front and the third window function at the rear is set according to the value in the frequency direction so that the value near the center in the frequency range of the spectrum set becomes large. A frequency component changing means for applying and changing the changing window function F and deleting the component of the central second window function;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the changed spectrum intensity to generate a modified acoustic frame;
Modified acoustic frame output means for sequentially outputting the generated modified acoustic frames;
An information embedding device for an acoustic signal, comprising: In claim 1,
The frequency component changing means sets a predetermined fixed value V as the intensity of the predetermined frequency band, and uses the fixed value V to set a predetermined value in the spectrum of the first window function and the third window function. An apparatus for embedding information in an acoustic signal, wherein the ratio of spectral intensity of the first window function and the third window function is changed by giving intensity. In claim 1 or claim 2,
The acoustic frame reading means reads the preceding acoustic frame and a predetermined number of samples in an overlapping manner, multiplies the entire read acoustic frame by a predetermined window function and passes it to the frequency conversion means,
The apparatus for embedding information in an acoustic signal, wherein the modified acoustic frame output means outputs the generated modified acoustic frame by connecting it with a preceding modified acoustic frame. In any one of Claims 1-3,
The telephone line bandwidth upper limit is 3.4 kHz,
The frequency component changing means has a predetermined frequency band whose upper limit is the telephone line band upper limit among predetermined frequency bands whose upper limit or lower limit is the upper limit of the telephone line band. An apparatus for embedding information in an acoustic signal, wherein a predetermined frequency band having a lower limit of a line band upper limit is set in a range of 3.4 kHz to 6.8 kHz. In claim 1,
The acoustic signal is a stereo acoustic signal composed of a time-series sample sequence of two left and right channels,
The sound frame reading means reads sound frames corresponding to the respective channels, and the frequency converting means performs frequency conversion on the sound frames of the respective channels, and the first window spectrum and the second window spectrum for each channel. , Generating a third window spectrum,
The frequency component changing means is configured to select the first window based on the value of the information array of the additional information to be embedded for three sets of spectrum sets corresponding to a predetermined frequency band extracted from the window spectrum of one channel. The spectral intensity of the other window to change the ratio of the spectral intensity of the function and the third window function, delete the component of the second window function, and supplement the component deleted by the change made in the one channel. Is to change the strength ratio between
The frequency inverse transform means generates a modified acoustic frame by performing frequency inverse transform on the frame spectrum including the changed spectrum intensity for each channel, and the modified acoustic frame output means is generated for each channel. A device for embedding information in an acoustic signal, wherein the modified acoustic frames are sequentially output. A method of embedding additional information in an inaudible state for an acoustic signal composed of a time-series sample sequence,
From the acoustic signal, an acoustic frame reading stage for reading a predetermined number of samples as an acoustic frame;
The first window function, the second window function, and the second window function, which are spectrums corresponding to the first window function, by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, respectively. A frequency conversion step of generating a second window spectrum that is a spectrum corresponding to, a third window spectrum that is a spectrum corresponding to the third window function;
A spectrum set corresponding to a predetermined frequency band whose upper limit or lower limit is the telephone line band upper limit is extracted from each generated window spectrum, and the extracted 3 is extracted based on the value of the information array of the additional information to be embedded. Among the set of spectrum sets, the ratio of the spectrum intensity of the first window function at the front and the third window function at the rear is set according to the value in the frequency direction so that the value near the center in the frequency range of the spectrum set becomes large. Applying and changing the changing window function F, and changing the frequency component to remove the component of the central second window function;
A frequency inverse transform step of performing a frequency inverse transform on each window spectrum including the changed spectrum intensity to generate a modified acoustic frame;
A modified sound frame output step of sequentially outputting the generated modified sound frames;
A method for embedding information in an acoustic signal, comprising:   The program for functioning a computer as an information embedding apparatus with respect to the acoustic signal in any one of Claims 1-5. An apparatus for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
An acoustic frame acquisition means for digitizing a predetermined section of the acoustic signal and acquiring an acoustic frame composed of a predetermined number of samples;
Frequency conversion is performed on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a spectrum corresponding to the third window function are spectra corresponding to the first window function. A frequency converting means for generating a third window spectrum;
A spectrum set corresponding to a predetermined frequency band whose upper limit or lower limit is a telephone line band upper limit is extracted from each generated window spectrum, and the total value of spectrum intensities for each spectrum set is obtained as a frequency range of the spectrum set. A code that calculates and applies a window function F that changes in accordance with the value in the frequency direction so that the value in the vicinity of the center of the signal becomes large, and outputs a predetermined code based on the ratio between the spectrum sets of the total value And
Additional information extracting means for extracting the additional information by converting the information array corresponding to the output code according to a predetermined rule;
An apparatus for extracting information from an acoustic signal, comprising: In claim 8,
The acoustic frame acquisition means changes a phase by moving a reference frame and a predetermined sample from a reference frame acquisition means for acquiring an acoustic frame composed of a predetermined number of samples as a reference frame from the acoustic signal. It is constituted by phase change frame setting means for setting a plurality of set sound frames as phase change frames,
The encoding means is based on the code determination parameter calculation means for calculating a code determination parameter based on the extracted spectrum set, and on the code determination parameter calculated in a past in-phase acoustic frame having a different reference frame, It is determined that one of the reference frame and the plurality of phase change frames has an optimal phase, and a predetermined code is determined based on the code determination parameter determined for the acoustic frame having the optimal phase. An apparatus for extracting information from an acoustic signal, characterized by comprising code output means for outputting. A method for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
An acoustic frame acquisition step of digitizing a predetermined section of the acoustic signal to acquire an acoustic frame composed of a predetermined number of samples;
Frequency conversion is performed on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a spectrum corresponding to the third window function are spectra corresponding to the first window function. A frequency conversion stage for generating a third window spectrum;
A spectrum set corresponding to a predetermined frequency band whose upper limit or lower limit is a telephone line band upper limit is extracted from each generated window spectrum, and the total value of spectrum intensities for each spectrum set is obtained as a frequency range of the spectrum set. A code that calculates and applies a window function F that changes in accordance with the value in the frequency direction so that the value in the vicinity of the center of the signal becomes large, and outputs a predetermined code based on the ratio between the spectrum sets of the total value Conversion stage,
An additional information extraction step of extracting the additional information by converting the information array corresponding to the output code according to a predetermined rule;
A method for extracting information from an acoustic signal.   A program for causing a computer to function as an apparatus for extracting information from an acoustic signal according to claim 8 or 9.

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