JP4713181B2 - Information embedding device for sound signal, device for extracting information from sound signal, and sound signal reproducing device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a device for embedding information in a sound signal which is compatible with both of monaural and stereo sound signals and easily extract the embedded additional information, and also to provide a device for extracting the information from the sound signal, and a device for reproducing the sound signal. <P>SOLUTION: The sound signal is separated according to frequency bands, and the states are made to vary to two low frequency bands responding to three values in total of two bit strings and a separation to be embedded, then, the states are synthesized with the high frequency components. Regarding a stereo sound signal, it is embedded in an L-channel signal, but signal deterioration is prevented by adding the components deleted for embedding processing to an R-channel. <P>COPYRIGHT: (C)2006,JPO&amp;NCIPI

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, and a character information providing service field linked to exhibition narration at museums and event venues.

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

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

In the methods described in Patent Documents 1 to 6, the amount of information that can be embedded is small, the sound quality is deteriorated to some extent, watermark information is lost by various signal processing, and watermark detection is difficult for analog copies. There is a problem. Therefore, the present applicant changes the ratio of the low frequency component of the acoustic signal having a plurality of channels according to the bit value of the attribute information, and embeds the attribute information, or changes the phase of the monaural signal to the bit value of the attribute information. A method of embedding attribute information (additional information) by changing it accordingly has been proposed (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 Japanese Patent Application No. 2005-5157 Japanese Patent Application No. 2005-13490

  However, the technique described in Patent Document 7 cannot deal with monaural data, and if monaural conversion is performed, additional information such as embedded attribute information disappears. Further, the extraction requires a stereo microphone and stereo acoustic signal processing, and it is difficult to reduce the size of the extraction device, and the extraction operation is difficult because of the use of a directional microphone.

  Further, in the method described in Patent Document 8, when additional information such as attribute information is extracted via analog, the bit detection accuracy is particularly bad at the time of microphone input, so a high-accuracy acoustic input system is required, and the extraction device There is a problem that the realization cost becomes high.

  Therefore, the present invention can deal with both monaural and stereo sound signals, and can easily extract the embedded additional information. It is an object to provide an extraction device and an acoustic signal reproduction device.

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 a stereo acoustic signal composed of a time-series sample sequence of two left and right channels, and from the acoustic signal, An acoustic frame reading unit that reads a predetermined number of samples from each channel as an acoustic frame; a frequency conversion unit that performs frequency conversion on the acoustic frame of each channel and generates a frame spectrum for each channel; and the generated Two sets of spectrum sets in two predetermined frequency ranges are extracted from the frame spectrum of one channel, and based on the value of the information array of the additional information to be embedded, the ratio between the frequency ranges of the sum of the intensities of the spectrum sets with changing the two corresponding to the frame spectra from other channel The spectral set in the frequency range and two sets of extraction, changing the ratio between the frequency range of the sum of intensities of the spectral set to complement the reduction ingredients by changes made in the one channel low-frequency a component changing unit, for each channel, performs inverse frequency transformation on the frame spectrum comprising said modified spectral set, and the frequency inverse conversion means for generating a modified acoustic frame, for each channel, the generated modified acoustic Provided is an information embedding device for an acoustic signal having modified acoustic frame output means for sequentially outputting frames.

  According to the present invention, the intensity ratio between two low frequency bands in the low frequency region of the acoustic signal is changed based on the information arrangement of the additional information to be embedded. And the embedded additional information can be easily extracted.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Information embedding device for acoustic signals)
FIG. 1 is a functional block diagram showing the configuration of an information embedding device for an acoustic signal according to the present invention. In FIG. 1, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a low frequency component changing means, 40 is a frequency inverse converting means, 50 is a modified acoustic frame output means, 60 is a storage means, and 61 is an acoustic signal storage. , 62 is an additional information storage unit, 63 is a modified acoustic signal storage unit, and 70 is an additional information reading means. The apparatus shown in FIG. 1 can deal with both a stereo sound signal and a monaural sound signal. Here, a case where processing is performed on a stereo sound signal will be described.

  The sound frame reading means 10 has a function of reading a predetermined number of samples as one frame from each channel of the original stereo sound signal to be embedded with additional information. The frequency conversion means 20 has a function of generating a frame spectrum by frequency-converting the frame of the acoustic signal read by the acoustic frame reading means 10 by Fourier transformation or the like. The low frequency component changing means 30 extracts two sets of spectrum sets corresponding to two predetermined frequency ranges from the generated frame spectrum, and based on the additional information extracted from the additional information storage unit 62, the spectrum of the low frequency intensity data It has a function to change the ratio between sets. The frequency reverse conversion means 40 has a function of generating a modified acoustic frame by performing frequency reverse conversion on a plurality of frame spectra including the changed low frequency intensity data. The modified sound frame output means 50 has a function of sequentially outputting the generated modified sound frames. The storage means 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded with additional information, an additional information storage unit 62 that is configured as a bit array and stores additional information embedded in the stereo acoustic signal, and an additional information It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after information is embedded, and stores various information necessary for other processing. The additional information reading means 70 has a function of extracting additional information from the additional information storage unit 62. The additional information is information that should be added to the sound information and embedded, and includes attribute information such as a title and artist name, and other information other than the attribute information. Each component shown in FIG. 1 is actually realized by installing a dedicated program in hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

(2. Processing operation of embedded device)
Next, the processing operation of the information embedding device for the acoustic signal shown in FIG. 1 will be described with reference to the flowchart of FIG. Here, a case where processing is performed on a stereo sound signal having two channels of L (left) and R (right) as sound signals will be described. FIG. 2 corresponds to processing of 1 byte of additional information. First, the additional information reading means 70 reads additional information from the additional information storage unit 62 in units of 1 byte (S101). Specifically, 1 byte is read into the register. Subsequently, the mode is set to the separation mode (S102). There are two types of modes: separation mode and bit mode. The delimiter mode indicates a mode for performing processing at a delimiter in units of 1 byte, and the bit mode indicates a mode for performing processing based on the value of each bit of 1 byte. When one byte is read from the additional information storage unit 62, the separator mode is always set immediately after that.

  Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as one acoustic frame from each of the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S104). The number of samples of one sound frame read by the sound frame reading means 10 can be set as appropriate, but is desirably about 4096 samples when the sampling frequency is 44.1 kHz. Therefore, the acoustic frame reading means 10 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames.

  Subsequently, the frequency conversion means 20 performs frequency conversion on each read sound frame to obtain a frame spectrum that is a spectrum of the sound frame (S105). 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. When performing Fourier transform, the left channel signal xl (i) and the right channel signal xr (i) (i = 0,..., N−1) are processed according to the following [Equation 1], and left The real part Al (j) and imaginary part Bl (j) of the conversion data corresponding to the channel and the real part Ar (j) and imaginary part Br (j) of the conversion data corresponding to the right channel are obtained.

[Formula 1]
Al (j) = Σ _{i = 0,..., N−1} xl (i) · cos (2πij / N)
Bl (j) = Σ _{i = 0,..., N−1} xl (i) · sin (2πij / N)
Ar (j) = Σ _{i = 0,..., N−1} xr (i) · cos (2πij / N)
Br (j) = Σ _{i = 0,..., N−1} xr (i) · sin (2πij / N)

  In [Expression 1], i is a serial number assigned to N samples in each acoustic frame, and takes an integer value of i = 0, 1, 2,... N−1. Further, j is a serial number assigned in order from the smallest value of the frequency value, and takes an integer value of j = 0, 1, 2,... N−1 similarly to i. When the sampling frequency is 44.1 kHz and N = 4096, if the value of j is different by one, the frequency will be different by 10.8 Hz. At this time, the acoustic signals xl (i) and xr (i) are each weighted with a window function (Hanning window) expressed by W (i) = 0.5−0.5 · cos (2πi / N). Multiply. Such a window function is used to reduce high-frequency noise generated by dividing the waveform into frequency components when performing Fourier transform, and between analysis intervals (corresponding to acoustic frames) when performing inverse Fourier transform. This is a well-known technique used to connect signal levels so as not to be discontinuous.

  By executing the processing according to the above [Equation 1], a frame spectrum in which the signal component of each acoustic frame is expressed by a spectrum corresponding to the frequency is obtained. Subsequently, the low frequency component changing unit 30 extracts two sets of spectra in a predetermined frequency range from the generated frame spectrum. Human hearing is known to be less sensitive to directionality for low frequency components up to about 200-300 Hz (Corona Corp., issued October 30, 1990, "Sound Engineering Course 1. Basic Acoustics"). Engineering, Acoustical Society of Japan ”p.247 (see FIGS. 9 and 26). Therefore, in this embodiment, the low frequency component is set to about 200 Hz or less, and the low frequency component is divided into two near 100 Hz and extracted as the first low frequency band and the second low frequency band. In the vicinity of frequencies of 100 Hz and 200 Hz, j corresponds to 10 and 20, respectively. Therefore, the real part Al (j), the imaginary part Bl (j), the real part Ar (j) calculated by the above [Equation 1], Of the imaginary part Br (j), those having j ≦ 10 as the first low frequency band and 11 ≦ j ≦ 20 as the second low frequency band are extracted.

Subsequently, the low-frequency component changing unit 30 extracts the real part Al (left channel) of the real part Al (j), the imaginary part Bl (j), the real part Ar (j), and the imaginary part Br (j). j), using the imaginary part Bl (j), the following sums E ₁ and (M1 + 1) to M2 (for example, M1 = 10, M2) of the low frequency intensity from j = ₁ to M1 by the following [Equation 2] = 20) The total value E ₂ of the low frequency intensity up to 20) is calculated.

[Formula 2]
E ₁ = Σj _{= 1,..., M1} {Al (j) ² + Bl (j) ² }
E ₂ = Σ _{j = M1 + 1,..., M2} {Al (j) ² + Bl (j) ² }

E ₁ and E ₂ calculated by the above [Equation 2] indicate the sum of the component intensities of the respective spectrum sets in the first low frequency band and the second low frequency band in the frame spectrum. Subsequently, it is determined whether or not the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev. The level lower limit value Lev is set to 2.0 when the maximum amplitude value of the acoustic signals xl (i) and xr (i) is normalized to 1 and M2 = 20. The reason why it is determined whether or not the combined values E ₁ and E ₂ are equal to or greater than the level lower limit value Lev is that if the signal strength is small, even if the signal is changed, the change cannot be detected on the extraction side. is there. Therefore, when both the sum values E ₁ and E ₂ are less than the level lower limit value Lev, the recording is not performed according to the bit value of the additional information, but is processed again from the first bit, so that the reading position is returned to the first bit, The mode is set to the separation mode (S106). On the other hand, when the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev, the mode is determined.

  When the mode is the separation mode, the low frequency component changing unit 30 performs a process for making the first low frequency band and the second low frequency band uniform (including both 0) (S108). Specifically, according to the following [Equation 3], processing for setting both L side to 0 is executed. In this case, the components of the first low frequency band and the second low frequency band on the R side are not equal.

[Formula 3]
E (j) = {Al (j) ² + Bl (j) ² + Ar (j) ² + Br (j) for both the first low frequency band (j = 1 to 10) and the second low frequency band (j = 11 to 20) ² } ^1/2
Al (j) = 0
Bl (j) = 0
Ar (j) ← Ar (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2
Br (j) ← Br (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2

  In the above [Expression 3], “←” in the fourth and fifth expressions indicates that the calculation result of the right side is substituted into the left side. By executing the processing according to the above [Equation 3], the low frequency component of the frame spectrum of the left channel is the same at “0” in both the first low frequency band and the second low frequency band. The pattern in which the first low frequency band and the second low frequency band are equal is information indicating the head position (separation) of the additional information. In the above [Equation 3], Al (j) = Bl (j) = 0 is set for both the first low frequency band and the second low frequency band, but it is possible to recognize the separation on the extraction side. Therefore, if the value is sufficiently small, it is not always necessary to set it to zero.

  On the other hand, when the mode is the bit mode, the low frequency component changing unit 30 determines the distribution between the first and second low frequency bands according to the bit value of the bit arrangement of the additional information extracted from the additional information storage unit 62. A process of changing whether the first low frequency band is dominant or the second low frequency band is dominant is performed (S107). Specifically, the first and second values are obtained by executing processing according to any of the following [Equation 4] and [Equation 5] according to the bit value that can take the first value and the second value. 2 Change the magnitude relationship between the low frequency bands. For example, when the first value is 1 and the second value is 0, when the bit value is 1, processing according to the following [Equation 4] is executed. In this case, no processing is performed for the second low frequency band (j = 11 to 20).

[Formula 4]
For the first low frequency band (j = 1 to 10), Al (j) = 0
Bl (j) = 0
E (j) = {Al (j) ² + Bl (j) ² + Ar (j) ² + Br (j) ² } ^1/2
Ar (j) ← Ar (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2
Br (j) ← Br (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2

  When the bit value is 0, processing according to the following [Formula 5] is executed. In this case, no processing is performed for the first low frequency band (j = 1 to 10).

[Formula 5]
For the second low frequency band (j = 11 to 20), Al (j) = 0
Bl (j) = 0
E (j) = {Al (j) ² + Bl (j) ² + Ar (j) ² + Br (j) ² } ^1/2
Ar (j) ← Ar (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2
Br (j) ← Br (j) · E (j) / {Ar (j) ² + Br (j) ² } ^1/2

  Whether the first low frequency band is dominant or the second low frequency band is executed according to each bit value of the bit array of the additional information by executing the process according to any one of [Formula 4] and [Formula 5]. Will be changed to one of the dominant patterns. Eventually, the low frequency component changing means 30 performs the processing based on [Equation 3] in the case of the separation mode in S108, and performs the processing based on [Equation 4] or [Equation 5] in S107 in the case of the bit mode. Become.

  Next, the frequency inverse transform means 40 performs a process of obtaining a modified acoustic frame by performing frequency inverse transform on the frame spectrum in which the ratio between the spectrum sets of the low frequency intensity data is changed (S109). As a matter of course, this frequency inverse transform needs to correspond to the method executed by the frequency transform unit 20 in S105. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the real part Al (j), the imaginary part Bl (j), and the real part Ar (j) of the right channel of the spectrum obtained by any one of [Formula 3] to [Formula 5]. Then, using imaginary part Br (j), processing according to the following [Equation 6] is performed to calculate xl ′ (i) and xr ′ (i).

[Formula 6]
xl ′ (i) = 1 / N · {Σj _{= 0,..., N-1} Al (j) · cos (2πij / N) −Σj _{= 0,..., N−1} B1 (j) · sin ( 2πij / N)} + {1-W (i)} · xl (i)
xr ′ (i) = 1 / N · {Σ _{j = 0,..., N−1} Ar (j) · cos (2πij / N) −Σ _{j = 0,..., N−1} Br (j) · sin ( 2πij / N)} + {1-W (i)} · xr (i)

  The terms “+ {1−W (i)} · xl (i)” in the first equation and “+ {1−W (i)} · xr (i)” in the second equation in the above [Equation 6] are When the frequency conversion is performed in the above [Equation 1], the signal component removed by multiplying by the window function W (i) is restored. By the above [Equation 6], each sample xl ′ (i) of the left channel and each sample xr ′ (i) of the right channel of the modified sound frame are obtained. The modified sound frame output unit 50 sequentially outputs the obtained modified sound frames to the output file (S110). When the processing for one acoustic frame is completed in this way, after the mode is set to the bit mode (S111), the additional information reading means 70 reads the next bit in the bit array of the additional information (S103). The above processing is executed over all samples of both channels of the acoustic signal. That is, a predetermined number of samples are read as sound frames, and when there are no more sound frames to be read from the sound signal (S104), the process ends. When the processing corresponding to each bit of the 1-byte data read in S101 is completed, the process returns from S103 to S101, and the next byte of the additional information is read. When the processing is completed for all the bytes of the additional information, the processing returns to the first byte of the additional information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

The change of the left channel signal by the above processing will be described with reference to FIG. In FIG. 3, the left-right direction is a time axis and is proportional to the number of samples. In addition, a large number of rectangles in the figure indicate the first low frequency band and second low frequency band components of the modified acoustic frame, the horizontal width indicates the number of samples (4096 in the present embodiment), and the vertical width indicates the intensity. Yes. FIG. 3A shows a good signal when there is no acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Equation 2] are less than the level lower limit value Lev, that is, for embedding additional information. The case is shown. FIG. 3B shows a signal that is not good for embedding additional information when there is an acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Equation 2] are less than the level lower limit value Lev. The case is shown.

For example, it is assumed that a 2-byte bit array in which the first byte is “11011100” and the second byte is “11000001” is embedded as additional information. First, at the head of each byte, the first low frequency band and the second low frequency band are set to be equal as information indicating a delimiter. This is obtained as a result of executing the processing in accordance with the above [Equation 3] in S108 by setting the separation mode in S102. In the example of FIG. 3 (a), since there is no acoustic frame in which the combined values E ₁ and E ₂ are less than the level lower limit value Lev, 1 byte is continuously obtained by the above [Expression 4] or [Expression 5]. Will be processed. This indicates that the loop from S103 to S111 was repeated 8 times in succession, and during that time, it did not pass through S106 and S108 because it was less than the level lower limit value. As shown in FIG. 3, when the bit value of the additional information is 1, a low frequency component exists on the first low frequency band side, and when the bit value of the additional information is 0, the low frequency component is on the second low frequency band side. There is a low frequency component. As can be seen from the above [Equation 4] and [Equation 5], in this case, the low frequency component of the other low frequency band is zero.

  In the example of FIG. 3B, there is an acoustic frame that is less than the level lower limit value Lev as a result of the process according to the above [Equation 2]. In this case, the above [Equation 3] is passed through S106 and S108. As a result of executing the processing according to, the first low frequency band and the second low frequency band are set to be in an equal state. In this case, since the reading position is returned to the first bit in S106, the same bit is read again. In the example of FIG. 3B, when “11011100” in the first byte is embedded, an acoustic frame less than the level lower limit value Lev appears at the time when the first two bits of “11” are processed, and the second time “11011”. ”Indicates that an audio frame less than the level lower limit value Lev appeared at the time of 5 bit processing, and finally 8 bit processing could be performed at the third time.

  However, when the processing according to FIG. 2 is performed, as shown in FIG. 3B, every time an acoustic frame having a low signal level of low frequency components appears, it is repeated from the beginning in units of 1 byte. Since it will be processed, efficiency is bad. Therefore, instead of FIG. 2, processing as shown in FIG. 4 may be performed.

  The processing shown in FIG. 4 differs greatly from the processing shown in FIG. 2 in that it has a continuous identification mode in addition to the separation mode and the bit mode, and has three modes, and an acoustic frame with a low signal level of low frequency components has appeared. Even in this case, the process is continued without returning to the top. Therefore, in the continuous identification mode, information for identifying whether the next bit is a new one starting from the beginning or continued because it is interrupted is recorded. In the process of FIG. 4, the additional information is processed in units of words instead of units of bytes. This is because, as shown in S106 of FIG. 2, there is no process for forcibly returning the reading position to the beginning, and it is easy to process in units other than bytes. The number of bits in one word can be set freely and can be set to 1 byte.

The processing in FIG. 4 has many parts in common with the processing in FIG. First, the additional information reading means 70 reads additional information from the additional information storage unit 62 in units of one word (S201). Thereafter, the mode is set to the separation mode (S202), and the acoustic frame reading means 10 reads the acoustic frames from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S204), and the frequency converting means 20 However, frequency conversion is performed on each read acoustic frame according to [Equation 1] to obtain a frame spectrum (S205), and the low frequency component changing means 30 extracts the real part Al (j) and imaginary part Bl. (J) Using the real part Ar (j) and imaginary part Br (j), the combined values E ₁ and E ₂ are calculated by [Formula 2], and the combined values E ₁ and E ₂ are the level lower limit value Lev. It is the same up to the point where it is judged whether it is above. However, in the process of FIG. 4, when the summed values E ₁ and E ₂ are less than the level lower limit value Lev, the mode is simply set to the separation mode (S206). On the other hand, when the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev, the mode is determined.

  When the mode is the separation mode, the low frequency component changing unit 30 performs the process of making the low frequency intensity the same in the first low frequency band and the second low frequency band in accordance with the above [Equation 3] (S208). On the other hand, when the mode is the bit mode, the low frequency component changing means 30 according to the above [Equation 4] and [Equation 5] according to the bit value of the bit array of the additional information extracted from the additional information storage unit 62, A process of changing the ratio between the first low frequency band and the second low frequency band of the low frequency intensity is performed (S207). Further, in S207, unlike S107 in FIG. 2, processing is performed even in the continuous identification mode. In the case of the continuous identification mode, when it is new, the distribution between the first low frequency band and the second low frequency band of the low frequency component is changed to the large value in the first low frequency band according to [Equation 4], and it is continued. In some cases, the distribution of the low-frequency component between the first low-frequency band and the second low-frequency band is changed to a large value by the second low-frequency band according to [Equation 5].

  Next, the frequency reverse conversion means 40 performs a process of performing frequency reverse conversion on the frame spectrum in which the ratio between the spectrum sets of the low frequency intensity data is changed to obtain a modified sound frame according to [Formula 6] (S209), The frame output means 50 sequentially outputs the obtained modified sound frames to the output file (S210). When the processing for one acoustic frame is completed in this manner, the mode is determined (S211). If the mode is the separation mode, the mode is set to the continuous identification mode (S212), and then the acoustic frame reading means 10 A frame is read (S204). On the other hand, when the mode is the bit mode or the continuous identification mode, after setting the mode to the bit mode (S213), the low frequency component changing means 30 reads the next bit in the bit array of the additional information (S203). Then, all the sound frames are processed, and when there are no more sound frames to be read from the sound signal (S204), the process ends.

A state of the change of the left channel signal by the processing according to FIG. 4 will be described with reference to FIG. In FIG. 5, as in FIG. 3, the horizontal direction is the time axis, and a large number of rectangles in the figure indicate the first low frequency band and second low frequency band components of the modified acoustic frame, and the horizontal width is the number of samples. The vertical width indicates the strength. FIG. 5A shows a case where there is no acoustic frame in which the sum values E ₁ and E ₂ calculated by the above [Equation 2] are equal to or higher than the level lower limit value Lev, as in FIG. 3A. FIG. 5 (b) shows a case where there is an acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Formula 2] are less than the level lower limit value Lev, as in FIG. 3 (b). Yes.

  For comparison with the processing of FIG. 2, the additional information to be embedded is also a 2-byte bit array in which the first byte is “11011100” and the second byte is “11000001”, as in FIG. As in FIG. 3, the first low frequency band and the second low frequency band are set to be uniform at the head of each byte as information indicating a delimiter. This is obtained as a result of executing the processing according to the above [Equation 3] in S208 by setting the separation mode in S202. Subsequently, before the processing corresponding to each bit of the additional information is performed, information indicating whether the information is new or continued is recorded. According to the process shown in FIG. 2, when there is an acoustic frame that is less than the level lower limit value Lev, the bit to be processed is returned to the beginning of each byte and processed, so all are new, but as shown in FIG. In the processing, even if there is an acoustic frame that is less than the level lower limit value Lev, the bit processed at that time is valid and continues from there, so whether the bit is new or continued It is necessary to record the information. Therefore, after recording the information indicating the break, information indicating whether it is new or continued is recorded. Specifically, the mode determination is performed in the separation mode state (S211), the continuous identification mode is set (S212), and the sound frame is extracted without reading the bits of the additional information (S204). Then, after the frequency conversion (S205), if new, the distribution between the first low frequency band and the second low frequency band of the low frequency component is changed to the first low frequency band by the process according to [Equation 4]. Is changed to large (S207).

  After recording the information indicating whether it is new or continued in this way, in order to make a mode determination in the state of the continuous identification mode (S211), the bit mode is set (S213), and the first bit is read from the register ( (S203), an acoustic frame is extracted (S204). In the example of FIG. 5A, since there is no acoustic frame that is less than the level lower limit value Lev, one byte is continuously processed by the above [Expression 4] or [Expression 5]. This indicates that the loop from S203 to S213 was repeated 8 times in succession, and during that time, it was assumed that the level was lower than the lower limit value Lev, and that it did not go through S206, S208, and S212.

  In the example of FIG. 5B, as a result of the processing according to the above [Equation 2], there is an acoustic frame that is less than the level lower limit Lev. In this case, the above [Equation 3] passes through S206 and S208. As a result of executing the processing according to, the first low frequency band and the second low frequency band are set to be in an equal state. In this case, since the separation mode is set in S206, information indicating whether it is new or continued is recorded via S212. In the example of FIG. 5B, when embedding “11011100” of the first byte, an acoustic frame less than the level lower limit value Lev appears when 1 bit of “1” of the first bit is initially processed. Therefore, after the information indicating the break is recorded, the information indicating the continuation is recorded, and the processing is continued from “1” of the second bit. Then, since “1011” from the second bit to the fifth bit has been processed, an acoustic frame less than the level lower limit value Lev has appeared, so after recording the information indicating the break, the information indicating the continuation is recorded. Then, processing is continued from “1” of the sixth bit.

  In the example of FIG. 5, the case where the additional information is recorded in units of bytes has been described for comparison with FIG. 3, but the process shown in FIG. 4 records information indicating whether it is new or continued. Additional information can be recorded in an arbitrary number of bits.

  In the above example, information indicating a delimiter is inserted in a fixed-length byte unit or a variable-length word unit, but it is also possible to insert information indicating a delimiter in bit units. In this case, when the acoustic frame reading means 10 extracts the acoustic frame, it extracts an overlapping acoustic frame that overlaps the preceding and following acoustic frames, performs frequency conversion on the overlapping acoustic frame according to [Equation 1], and , [Equation 3] is performed to equalize the low frequency components of the first low frequency band and the second low frequency band. The overlapping sound frames are set so that half of the samples overlap with the preceding and following sound frames. For example, when the preceding sound frame is sample number 1 to 4096 and the subsequent sound frame is sample number 4097 to 8192, the overlapping sound frame set between this is sample number 2049 to 6144. Similarly, for all sections of the acoustic signal, a duplicate acoustic frame is read, and processing for equalizing the low frequency components of the first low frequency band and the second low frequency band is performed.

  As described above, when the overlapping acoustic frame is set and the process of equalizing the low frequency components of the first low frequency band and the second low frequency band is performed, in order to reflect this in the modified acoustic signal, 1. Perform frequency inverse transform on the processed overlapping frame spectrum to equalize the low frequency components of the low frequency band and the second low frequency band to obtain a modified overlapping acoustic frame, and further perform processing to connect with the acoustic frame There is a need. In this case, the signal component removed from the acoustic frame by multiplying by the window function W (i) can be compensated by connecting it with the signal of the overlapping acoustic frame. “+ {1−W (i)} · xl (i)” and “+ {1−W (i)} · xr (i)” in the second equation are not necessary. Therefore, in this case, the frequency inverse transformation for the overlapping frame spectrum and the frequency inverse transformation (S109, S209) for the frame spectrum are processed according to the following [Equation 7] instead of the above [Equation 6].

[Formula 7]
xl ′ (i) = 1 / N · {Σj _{= 0,..., N-1} Al (j) · cos (2πij / N) −Σj _{= 0,..., N−1} B1 (j) · sin ( 2πij / N)}
xr ′ (i) = 1 / N · {Σ _{j = 0,..., N−1} Ar (j) · cos (2πij / N) −Σ _{j = 0,..., N−1} Br (j) · sin ( 2πij / N)}

  By the above [Expression 7], each sample xl ′ (i) of the left channel and each sample xr ′ (i) of the right channel of the modified acoustic frame and the modified overlapping acoustic frame are obtained. In S110 and S210, the modified sound frame output means 50 sequentially connects the obtained modified sound frame and the modified overlapping sound frame and outputs them to the output file. As described above, when the sound frame and the overlapping sound frame are read from the sound signal by the sound frame reading means 10, the sound frame and the overlapping sound frame are processed so that the same sample is included in duplicate. Therefore, the acoustic frame output means 50 records the sample read in duplicate in the output file as the sum of the values of the samples.

  Of the left channel of the modified acoustic signal obtained in this way, for the portion in which the additional information is embedded, the low frequency component exists equally in the first low frequency band and the second low frequency band, Alternatively, there are only three distributions of uneven distribution in either the first low frequency band or the second low frequency band. However, since the high frequency component remains the original acoustic signal, it has various distributions based on the setting of the producer. Further, as shown in the above example, when a stereo sound signal is used, the low frequency component changed in the left channel is apparent from the processing of [Expression 3] to [Expression 5]. In addition, it is always added to the low frequency component of the right channel. Therefore, since the right channel supplements the deleted component in the left channel, there is no signal degradation when viewed as both channels as a whole. Human auditory senses directionality with respect to high-frequency components, but it is difficult to sense directionality with respect to low-frequency components. Therefore, even if the low frequency component is biased to one side, it will be heard as if it is a normal acoustic signal for the listener.

(3. 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. Any microphone can be used, as long as it can detect a low-frequency component, whether it is monaural omnidirectional or stereo directional. Even if it is stereo-directional, only one channel needs to be used. The reference frame acquisition unit 110 has a function of reading an audio frame composed of a predetermined number of samples as a reference frame from the input digital monaural audio signal (or one channel of a stereo audio signal). The phase change frame setting means 120 has a function of setting, as a phase change frame, an acoustic frame whose phase has been changed by moving the reference frame and a predetermined sample at a time. The frequency conversion means 130 has the same function as the frequency conversion means 20 shown in FIG. The code determination parameter calculation means 140 extracts each low frequency intensity data corresponding to a predetermined frequency or less from the generated frame spectrum, and adds the low frequency intensity data for each of the first low frequency band and the second low frequency band. A function of calculating values E ₁ and E ₂ , using the combined values E ₁ and E ₂ as code determination parameters, and determining a predetermined state based on the ratio of the code determination parameters E ₁ and E ₂ is doing.

  The code output means 150 determines what is determined to be the optimum phase from the acoustic frames (reference frame and phase change frame) corresponding to one reference frame, and selects a code corresponding to the state of the acoustic frame. It has a function to output. The additional information extraction unit 160 has a function of converting the ternary array, which is a set of codes output by the code output unit 150, according to a predetermined rule and extracting it as meaningful additional information. The acoustic frame holding means 170 is a buffer memory that can hold two consecutive reference frames. Each component shown in FIG. 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.

(4. 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, HL2, the frequency band balance table, and the phase determination table are initialized. These will be described. The average code levels HL1 and HL2 are the low frequencies calculated by the above [Equation 2] for an acoustic frame (hereinafter referred to as an effective frame) that is determined to have embedded binary values corresponding to bit values. sum E _1, the average value of E ₂ components, namely, which is given by the average value of the sum E _1, E ₂ in the past valid frame, the initial value is, the level limit is also used in the above embedding device The value Lev is set. The frequency band balance table is actually composed of two types of correction coefficients C1 (j) and C2 (j), and the initial value is C1 (j) = C2 ( j) = 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.

  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 (S301). Specifically, the reference frame is extracted and read into the acoustic frame holding unit 170. The number of samples of one acoustic frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the acoustic frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples as reference frames. The acoustic frame holding means 170 can store two reference frames as described above, and when a new reference frame is read, the old reference frame is discarded. Therefore, the sound frame holding means 170 always stores two reference frames (continuous 8192 samples).

  The acoustic frame processed by the embedding device can be divided into a reference frame that is set adjacently without interruption from the head, and a phase change frame in which the phase is changed. For the reference frame, after setting the first reference frame from sample number 1 to sample number 4096, the next reference frame is sample number 4097 to sample number 8192, and the next reference frame is sample number 8193 to sample number 12288. It is set without interruption. Then, for each reference frame, five phase change frames moved by 1/6 frame (about 683 samples) are set. For example, for the first reference frame, five phase change frames configured by 4096 samples starting from sample numbers 683, 1366, 2049, 2732, and 3413 are set.

  Subsequently, the frequency conversion unit 130 and the code determination parameter calculation unit 140 determine embedded information from each read sound frame and output a corresponding code (S302). The format of the information to be output is a binary format corresponding to the bit value on the embedding side and a ternary format of values input as delimiters.

  Details of the code determination process in step S302 will be described with reference to the flowchart of FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain a frame spectrum (S401). This process is the same as the process in the frequency conversion means 20 shown in FIG. However, since only the left channel is used for extraction, processing according to the above [Equation 1] is performed to obtain real part Al (j) and imaginary part Bl (j) of the conversion data corresponding to the left channel. . At this time, the left channel acoustic signal xl (i) is multiplied by a window function (Hanning window) expressed by W (i) = 0.5−0.5 · cos (2πi / N) as a weight. This is performed in the same manner as the frequency conversion means 20.

By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, the code determination parameter calculation unit 140 calculates the average code levels HL1 and HL2 (S402). Specifically, v1 is an integrated value of the sum E ₁ for acoustic frame is determined to be large past the first low frequency band, the number of sound frames is determined to be large past first low frequency band by dividing in a certain n1 calculates HL1, the sum is the accumulated value of the E ₂ v2 for acoustic frame that is determined to be large past the second low frequency band, a first low frequency band past is large determined HL2 is calculated by dividing by n2, which is the number of sound frames that have been performed. Therefore, the average code levels HL1 and HL2 are the average value of the sum of the low frequency intensity data of the acoustic frames determined to have a large corresponding low frequency band in the past.

Furthermore, the code determination parameter calculation unit 140 extracts each low frequency intensity data in a predetermined frequency range from the generated frame spectrum. The frequency range to be extracted needs to correspond to the embedding device. Therefore, here, low frequency intensity data having a frequency of 200 Hz or less is extracted, and the real part Al (j) and imaginary part Bl of the left channel calculated by the above [Equation 1] as in the case of the embedding device. Among (j), those with j ≦ 20 are extracted. Then, the code determination parameter calculation unit 140 corrects the frequency band balance (S403). Specifically, using the extracted real part Al (j) and imaginary part Bl (j), the intensity El (j) for each frequency component is calculated by the following [Equation 8] and the above [Equation 2], the combined value E ₁ of the low frequency intensities of the first low frequency band from j = _{1 to} M1 (for example, M1 = 10), and the second low frequency band from j = M1 + _{1 to} M2 (for example, M2 = 20). calculating the sum E ₂ of the low-frequency intensity as a code decision parameter. For the second low frequency band, E ₂ ′ is calculated using the following [Equation 9].

[Formula 8]
El (j) = Al (j) ² + Bl (j) ²

[Formula 9]
E ₂ '= E ₂ · Σ _{j = M1 + 1,..., M2} {C1 (j) / C2 (j)}

In the above [Equation 9], C1 (j) and C2 (j) are correction coefficients for frequency band balance, and “Σj _{= M1 + 1,..., M2} {C1 (j) / C2 (j)}” Is multiplied by the low frequency intensity of the second low frequency band to correct the frequency band balance. _{Conversely, "Σ j = 1, ...} , M1 {C2 (j) / C1 (j)}" to the low-frequency intensity of the first low frequency band similar effects to calculate the E ₁ 'by multiplying the obtained It is done.

  Subsequently, the code determination parameter calculation unit 140 initializes the candidate code table (S404). 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 sign determination parameter calculation unit 140 determines whether or not the sum value E ₁ of the low frequency intensity in the second low frequency band and the sum value E ₂ ′ of the low frequency intensity in the second low frequency band are each equal to or greater than a predetermined value. Is determined (S405). Specifically, 1/10 of the average code levels HL1 and HL2 are set as the predetermined values, respectively. When the total value E ₁ is less than one-tenth of the average code level HL1 and the total value E ₂ ′ is less than one-tenth of the average code level HL2, the code determination parameter calculation unit 140 uses the delimiter information (S409).

On the other hand, when either one of the total values E ₁ and E ₂ ′ is equal to or greater than a predetermined value, the code determination parameter calculation unit 140 performs the comparison determination of the calculated code determination parameters E ₁ and E ₂ ′ as follows. This is performed according to [Formula 10] (S406), and a code corresponding to the comparison result is output.

[Formula 10]
When E ₂ ′ / E ₁ > 2, the second low frequency band is large. When E ₁ / E ₂ ′> 2, the first low frequency band is large E ₂ ′ / E ₁ ≦ 2 and E ₁ / E ₂ If '≤2, both frequency bands are equal

The code determination parameter calculation unit 140 outputs a ternary code according to the determination result for each acoustic frame. That is, when it is determined that the first low frequency band is large, the first bit value (for example, “1”) is output (S407), and when it is determined that the second low frequency band is large, the second bit value is output. The bit value (for example, “0”) is output (S408), and if both frequency bands are determined to be equal, a code indicating delimiter information is output (S409). If it is determined in S406 that the first low frequency band is large, E ₁ is equal to or higher than HL1, and if it is determined that the second low frequency band is large, E ₂ ′ is equal to or higher than HL2. If these conditions are not satisfied, a code indicating delimiter information is output (S409).

  When it is determined that the first low frequency band is large and the first bit value is output (S327), or when it is determined that the second low frequency band is large and the second bit value is output (S328) Further, the phase determination table S (p) is updated according to the following [Equation 11] (S410).

[Formula 11]
When the first low frequency band is large, S (p) ← S (p) + E ₁ / E ₂ ′
When the second low frequency band is large, S (p) ← S (p) + E ₂ ′ / E ₁

Subsequently, the code determination parameter calculation unit 140 saves the candidate for the optimum phase in the candidate code table (S411). Specifically, the value of the phase number p that maximizes the value of S (p) recorded in the phase determination table, the code of any one of the three values determined in S407 to S409, and the sound frame The values of E ₁ (j) and E ₂ (j) corresponding to the low frequency component calculated by the above [Equation 8] are stored in the candidate code table as candidates for the optimum phase.

  Subsequently, it is determined whether or not the processing corresponding to all the phase numbers p has been completed (S412). This determines whether all phase change frames have been processed for a certain reference frame. In the present embodiment, since p takes a value from 0 to 5, if 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 S405. 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. (S413).

Returning to the flowchart of FIG. If the code corresponding to the bit value is output as a result of the processing in S302, the parameter of the average code level is updated (S303). Specifically, the average code level HL1, HL2 the molecule when calculating the integrated value v1, respectively summed value v2 E _1, plus E ₂ updates the integrated value v1, v2, the number of frames comprising the denominator The number of frames n1 and n2 is updated by adding 1 to n1 and n2, respectively. Subsequently, the bit value corresponding to the output code is stored in the buffer (S304). Subsequently, the bit counter is counted up (S305). Then, it is determined whether the bit counter is 8 bits or more (S306). As a result, when the bit counter is 8 bits or more, since the bit value for 1 byte is stored in the buffer, the additional information extraction unit 160 outputs the data for 1 byte in the buffer. (S307). On the other hand, if a value corresponding to the delimiter information is output as a result of the processing in S302, the frequency band balance table is updated (S308). Specifically, C1 (j) and C2 (j) constituting the frequency band balance table are updated by the following [Equation 12].

[Formula 12]
C1 (j) ← C1 (j) + E ₁ (j)
C2 (j) ← C2 (j) + E ₂ (j)

Note that E ₁ (j) and E ₂ (j) in the above [Equation 12] are recorded in the candidate storage table corresponding to the acoustic frame determined to be the optimum phase in S413. .

  Then, the bit counter is initialized to 0 (S309). By executing the processing shown in FIG. 7 for each acoustic frame, additional information is extracted. If it is determined in S301 that all sound frames have been extracted, the process ends.

  In the processing of S307, the additional information extracting unit 160 first indicates that the first low frequency band and the second low frequency band are equal among the ternary codes output by the code determination parameter calculating unit 140. A bit array is created by using a code as a delimiter position, the next code as the head, a code having a large first low frequency band and a code having a large second low frequency band corresponding to the bit value. Subsequently, this bit arrangement is converted according to a predetermined rule and extracted as meaningful additional information. As the predetermined rule, various rules can be applied as long as the information intended by the person who embeds the information can be recognized by the person who has received it. As a rule. In other words, the additional information extraction unit 160 recognizes the code output from the code output unit 150 as determined by the code determination parameter calculation unit 140 in units of 1 byte (8 bits), and character information according to the set code system. Recognize The character information thus obtained is displayed and output on a screen of a display device (not shown).

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

  In the above processing, in order to extract additional information accurately in the extraction device, processing for correcting the phase, processing for correcting the balance of the intensity of the low frequency components of the first low frequency band and the second low frequency band, and invalidity A process of correcting the lower limit threshold value for determining the frame is performed. Next, supplementary explanation will be given for these three correction processes.

(5. About phase correction processing)
As described above, at the time of extraction, it is not always possible to read an acoustic signal corresponding to the acoustic frame embedded at the time of embedding. Therefore, the phase of the acoustic frame is shifted and read in a plurality of ways (six in this embodiment), the optimum phase is determined, and a code corresponding to the acoustic frame specified by the phase is output. Yes. For example, 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 S404, S410, S411, S412, and S413.

(6. Low frequency band balance correction processing)
When the analog system is used, the signals of the first low frequency band and the second low frequency band of the acoustic signal influence each other, the balance between the two frequency bands is lost, and an erroneous determination is often made on the extraction side. Therefore, processing for correcting the balance between both frequency bands is performed using the integrated value for each frequency band corresponding to the past acoustic frame. This frequency band balance correction process is performed centering on the processes in S403 and S308.

(7. Lower limit correction process)
When the signal level is small, the magnitude of the intensity for each frequency band cannot be determined, and erroneous determination is often made on the extraction side. Therefore, frames whose low frequency intensities E ₁ and E ₂ ′ are equal to or lower than a predetermined threshold are determined to be invalid frames, but the threshold at this time is set as the low frequency intensities of past effective frames. Correction processing is performed using the integrated value. 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 threshold correction process is performed centering on the processes in S402 and S303.

  The flowchart in FIG. 7 corresponds to the additional information recorded on the embedding side in byte units. When additional information is recorded in units of words on the embedding side, processing according to the flowchart of FIG. 9 is performed. First, as in S301 of FIG. 7, the reference frame acquisition unit 110 reads a predetermined number of samples as one reference frame from each channel of the stereo audio signal input from the audio signal input unit 100 (S501).

  Subsequently, as in S301 of FIG. 7, the frequency conversion unit 130 and the code determination parameter calculation unit 140 specify the acoustic frame having the optimum phase from the reference frame or the phase change frame using each read reference frame, A code corresponding to the information embedded in the sound frame is output (S502). The details of the processing in S502 are as shown in FIG. 8 as in S302.

  If a code corresponding to the bit value is extracted as a result of the processing in S502, the parameter of either the average code level HL1 or HL2 is updated as in S303 (S503). Subsequently, the mode is determined (S504). Two modes, a delimited mode and a bit output mode, are prepared. If it is in the bit output mode, the bit value is stored in the buffer (S509). Subsequently, the bit counter is counted up (S510). On the other hand, if the result of determination in S504 is that the mode is the separation mode, it is further determined whether the extracted code means new or continuation (S505). As a result, if it is new, it means that one word has been completed immediately before, and the additional information extracting means 160 outputs the data for one word recorded in the buffer (S506). Then, the bit counter is initialized to 0 (S507). Further, the mode is set to the bit output mode (S508). If it is determined in S505 that the operation is to be continued, the value should be output to the bit in the buffer, so only the processing for setting the bit output mode is performed. If a code corresponding to the delimiter information is extracted in S502, the frequency band balance table is updated as in S308 (S511). Subsequently, the mode is set to the delimiter mode in order to extract information on whether it is new or continued from the next sound frame (S512). By executing the processing shown in FIG. 9 for each reference frame, additional information is extracted. If it is determined in S501 that all the reference frames have been extracted, the process ends.

(8. Sound signal reproduction device)
Next, the acoustic signal reproducing apparatus according to the present invention will be described. FIG. 10 is a block diagram showing an embodiment of an acoustic signal reproduction device according to the present invention. In FIG. 10, reference numeral 200 denotes an acoustic frame reading unit, 210 denotes an additional information extraction / display unit, 240 denotes a reproduction frame input unit, 250 denotes a reproduction frame storage unit, 260 denotes a sound device driver, 261 denotes a sound device, and 262 denotes a timer. The sound signal reproducing apparatus according to the present invention can reproduce either a monaural sound signal or a stereo sound signal, but the apparatus shown in FIG. 10 will be described as reproducing a stereo sound signal.

  Similar to the acoustic frame reading means 10 shown in FIG. 1, the acoustic frame reading means 200 has a function to frame and read a predetermined number of samples from each channel of the stereo acoustic signal. Each sound frame read by the sound frame reading means 200 is processed in two systems of the additional information extraction / display means 210 and the reproduction frame input means 240. More specifically, the additional information extraction / display unit 210 includes the phase change frame setting unit 120, the frequency conversion unit 130, the code determination parameter calculation unit 140, the code output unit 150, and the additional information extraction unit of the extraction apparatus illustrated in FIG. The means 160, the sound frame holding means 170, and the means for displaying and outputting the additional information extracted by the additional information extracting means 160 are configured.

  The reproduction frame input unit 240 has a function of inputting each audio frame read by the audio frame reading unit 200 to the reproduction frame storage unit 250. However, the reproduction frame input unit 240 not only simply inputs an audio frame, but also has a function of controlling the input of an audio frame when the reproduction frame storage unit 250 is not empty, as will be described later. . The reproduction frame accumulating means 250 has a plurality of buffer memories for accumulating the sound frames, and the sound frames accumulated in these buffer memories are input into the FIFO (first in first out) system, that is, first. It has a function to process information in a way that goes out first. That is, the playback frame storage unit 250 has a function of storing the sound frames input from the playback frame input unit 240 in the input order and passing them to the sound device driver 260 in that order. The sound device driver 260 has a function of driving the sound device 261 to reproduce sound frames, and the sound device 261 has a function of D / A converting the sound frames that are digital data and reproducing them as sound. Have. That is, the sound device driver 260 and the sound device 261 function as sound frame reproduction means. The timer 262 is a timer used for timing the reproduction of the acoustic signal by the sound device and the reproduction of the acoustic signal of the external device, and is shared with a timer that performs time management in the computer.

  In the acoustic signal reproduction apparatus shown in FIG. 10, as described above, there are two systems, namely, a system for extracting and displaying additional information in the additional information extraction / display means 210 and a system for reproducing sound by the reproduction frame input means 240 to the sound device driver 260. A system exists. When processing a monaural sound signal, since there is only one channel in the sound signal and additional information is embedded in that channel, both systems process the signal of one channel. On the other hand, when processing a stereo sound signal, the sound reproduction system processes the signal of two channels and performs stereo reproduction. However, the additional information extraction system only processes the signal of one channel in which the additional information is embedded. Will be processed.

(9. Processing operation of playback device)
Next, the processing operation of the acoustic signal reproducing device shown in FIG. 10 will be described. When the recording device on which the modified acoustic signal in which the additional information is embedded by the embedding device shown in FIG. 1 is recorded is reproduced on the reproducing device, first, the acoustic frame reading means 200 starts from each channel of the stereo acoustic signal read from the recording medium. Each of a predetermined number of samples is read as one acoustic frame. The number of samples of one sound frame read by the sound frame reading means 200 needs to be the same as that set by the sound frame reading means 10 shown in FIG. Therefore, in the present embodiment, the acoustic frame reading means 200 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames. Further, when the sound frame reading means 10 shown in FIG. 1 reads the sound frame so that the sample is overlapped with the sound frame adjacent to the sound frame, the sound frame reading means 200 is also adjacent to the sound frame. Read so that the sound frame and the sample overlap. In this case, the number of samples to be duplicated is the same as the number of samples duplicated by the acoustic frame reading means 10 and is 2048 samples, which is half of the 4096 samples constituting one acoustic frame. Even if the acoustic frame is read in duplicate, only the non-overlapping acoustic frame is passed to the additional information extraction / display unit 210. Further, the sound frame passed to the additional information extraction / display unit 210 is only for the left channel in which the additional information is embedded.

  Subsequently, the additional information extraction / display unit 210 performs the same processing as that of the extraction apparatus shown in FIG. 6 using each read sound frame as a reference frame, extracts the additional information, and displays and outputs it on the screen.

  On the other hand, the sound frame read by the sound frame reading means 200 is accumulated in the reproduction frame accumulation means 250 by the reproduction frame input means 240. If at least one frame is stored in the playback frame storage unit 250, the playback process by the sound device driver 260 is started. This playback process is performed by the sound frame reading unit 200, the additional information extraction display unit 210, and the playback frame input unit 240. Processing time is overwhelmingly longer than processing. (In other words, the additional information extraction and display unit 210 is designed to be overwhelmingly faster than the reproduction process.) Therefore, in the present embodiment, the reproduction frame accumulation unit 250 can accumulate up to four frames. As shown in FIG. 10, the sound frame reading means 200 continuously reads the next sound frame, and the playback frame input means 240 can store up to four sound frames in the playback frame storage means 250. Yes. When four frames are accumulated in the reproduction frame accumulation unit 250, the sound frame reading unit 200 and the reproduction frame input unit 240 are in a standby state until a space is generated in the reproduction frame accumulation unit 250. Accordingly, four frames are always stored in the playback frame storage unit 250, and the sound device driver 260 plays back the first acoustic frame among the acoustic frames stored in the playback frame storage unit 250. Specifically, the sound device 261 D / A converts the sound frame data and outputs it to the speaker. The acoustic frame that has been acoustically reproduced is deleted from the reproduction frame storage means 250.

  When the sound frame is deleted and there is room in the reproduction frame storage means 250, the sound frame is input from the reproduction frame input means 240 to the reproduction frame storage means 250. As a result, the reproduction frame storage means 250 stores the maximum capacity again. In reality, the read sound frame is input into the reproduction frame accumulating unit 250 when the CPU functions as the reproduction frame input unit 240. The reproduction frame input unit 240 not only simply inputs the audio frame to the reproduction frame storage unit 250 but also if the reproduction frame storage unit 250 has no free space, the reproduction frame storage unit 250 displays the audio frame reading unit 200 and the additional information extraction / display unit 210. On the other hand, a message for interrupting the process is sent to control the input of the sound frame to the reproduction frame storage means 250.

  On the other hand, the sound device 261 sequentially reproduces the first acoustic frame among the acoustic frames accumulated in the reproduction frame accumulation unit 250. At this time, each time the sound device 261 finishes the sound reproduction of one sound frame, the sound device 261 allows the sound frame reading means 200, the additional information extraction / display means 210, and the reproduction frame input means 240 to execute each process. Send.

  Here, the outline of the processing of the sound playback side in the playback apparatus, that is, the playback frame input means 240, the playback frame storage means 250, and the sound device driver 260 is organized and shown in the flowchart of FIG. First, the playback frame input means 240 searches for a free buffer memory in the playback frame storage means 250 (S601). If there is no vacant buffer memory, a message for interrupting the process is sent to the sound frame reading means 200 to wait for reception of a reproduction end message from the sound device 261 (S602). If there is a playback end message from the sound device driver 261, the playback end buffer is set as an empty buffer by deleting from the buffer memory storing the sound frame that has been played back (S603). Since the playback end message from the sound device 261 is simultaneously transmitted to the sound frame reading means 200, the additional information extraction / display means 210, and the playback frame input means 240, the sound frame reading means 200 and the playback frame input means 240 perform processing. The process is resumed (S604). Subsequently, the sound frame is stored in an empty buffer memory (S605). On the other hand, the sound device 261 always searches the buffer memory in the reproduction frame storage means 250 (S606), and when there is an acoustic frame, reproduces the acoustic frame (S607). Waiting for the reproduction of one acoustic frame (S608), when the reproduction is completed, a reproduction end message is transmitted to the acoustic frame reading means 200 and the reproduction frame input means 240 (S609).

  The sound signal reproduction device shown in FIG. 10 performs sound reproduction processing on the sound signal in the same manner regardless of whether or not the additional information is embedded. Therefore, the sound signal in which the additional information is not embedded is also reproduced as it is. Of the acoustic signal, the portion in which the additional information of the left channel is embedded is modified. However, since the high frequency component remains the original acoustic signal, it has various distributions based on the setting of the producer. As described above, when a stereo sound signal is used, the low-frequency component changed in the left channel is always changed to the low-frequency component in the right channel by the processing of [Formula 3] to [Formula 5]. Since it is added, there is no signal degradation when viewed as both channels as a whole. Therefore, it can be heard as normal sound signals.

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

  FIG. 12 is a conceptual diagram when embedding additional information according to the present invention in a stereo sound signal and a monaural sound signal. FIG. 12A shows a stereo sound signal, and FIG. 12B shows a monaural sound signal. In the example of FIG. 12, the low-frequency component for three acoustic frames is represented by a waveform, and the bit string to be embedded is a continuous bit string of “0” and “1” in the additional information, with a break between them. It is supposed to be inserted.

  In the case of a stereo sound signal, embedding is performed on the left channel (L-ch) signal. As shown in FIG. 12A, after frequency conversion, signal separation is performed, and bit embedding processing is performed. Specifically, bit embedding is performed as a result of the processing of [Formula 3] to [Formula 5]. Here, as described above, [Equation 4] is used when “0” is embedded, [Equation 3] when the separation mode is used, and [Equation 5] when “1” is embedded. Therefore, after the bit embedding process, the first (leftmost) acoustic frame has a signal component of the first low frequency band of 0 (represented by no waveform in the figure), and the central acoustic frame has the first low frequency band. The signal components in the second low frequency band are both 0, and the signal component in the second low frequency band is 0 in the last (rightmost) sound frame. At this time, as is clear from the contents of [Equation 3] to [Equation 5], the signal component from which the left channel signal is deleted is added to the right channel (R-ch) signal. Therefore, as shown in the lower part of FIG. 12A, the low-frequency component of the right channel signal is increased. After bit embedding processing, signal synthesis including high frequency components is performed, and then frequency inverse conversion is performed to obtain a modified acoustic signal.

  In the case of a monaural sound signal, processing is performed as shown in FIG. 12 (b), but as shown in the upper part of FIG. 12 (a), processing similar to the left channel of the stereo sound signal is performed. become.

It is a functional block diagram of an information embedding device for an acoustic signal. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. FIG. 3 shows how a low-frequency component is changed by processing according to FIG. 2. It is a flowchart which shows the process outline | summary by the other method of the apparatus shown in FIG. FIG. 5 shows how the low frequency component changes due to the processing according to FIG. 4. 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 S302 of FIG. It is a flowchart which shows the process outline | summary by the other method of the apparatus shown in FIG. It is a functional block diagram of the acoustic signal reproducing device according to the present invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a conceptual diagram of the embedding process of the additional information by this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Sound frame reading means 20 ... Frequency conversion means 30 ... Low frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified sound frame output means 60 ... Storage means 61 ... -Acoustic signal storage unit 62 ... Additional information storage unit 63 ... Modified acoustic signal storage unit 70 ... Additional information reading means 100 ... Acoustic signal input means 110 ... Reference frame acquisition means 120 ... Phase change frame setting means 130: Frequency conversion means 140 ... Code determination parameter calculation means 150 ... Code output means 160 ... Additional information extraction means 170 ... Acoustic frame holding means 200 ... Acoustic frame Reading means 210... Additional information extraction and display means 240... Playback frame input means 250... Playback frame storage means 260. Device driver 261 ... Sound device 262 ... Timer

Claims (8)

A device for embedding additional information in an inaudible state with respect to a stereo sound signal composed of a time-series sample sequence of two left and right channels ,
From the acoustic signal, acoustic frame reading means for reading a predetermined number of samples from each channel as an acoustic frame;
Frequency conversion means for performing frequency conversion on the acoustic frame of each channel and generating a frame spectrum for each channel ;
Two sets of spectrum sets in two predetermined frequency ranges are extracted from the generated frame spectrum of one channel, and the frequency of the sum of the intensities of the spectrum sets is extracted based on the value of the information array of the additional information to be embedded. While changing the ratio between the ranges, extract two sets of spectrum sets of two corresponding frequency ranges from the frame spectrum from the other channel to supplement the components reduced by the changes made in the one channel Low frequency component changing means for changing the ratio between the frequency ranges of the sum of the intensities of the spectrum set ;
For each channel, frequency inverse transform means for performing frequency inverse transform on the frame spectrum including the modified spectrum set to generate a modified acoustic frame;
Modified sound frame output means for sequentially outputting the generated modified sound frame for each channel ;
An information embedding device for an acoustic signal, comprising: In claim 1,
The acoustic frame reading means reads an adjacent 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 is configured to output the generated modified acoustic frame by connecting it to an adjacent modified acoustic frame. In claim 1 or claim 2,
The low frequency component changing means, information embedding device for the acoustic signal, characterized in that the spectral set of predetermined frequency range, and extracts from the two frequency ranges do not overlap with each other in a low frequency region of less than 200 Hz. It is a method of embedding additional information in an inaudible state with respect to a stereo sound signal composed of a time series sample sequence of two left and right channels ,
From the acoustic signal, an acoustic frame reading step of reading a predetermined number of samples from each channel as an acoustic frame;
A frequency conversion step of performing frequency conversion on the acoustic frames of each channel and generating a frame spectrum for each channel ;
Two sets of spectrum sets in two predetermined frequency ranges are extracted from the generated frame spectrum of one channel, and the frequency of the sum of the intensities of the spectrum sets is extracted based on the value of the information array of the additional information to be embedded. While changing the ratio between the ranges, extract two sets of spectrum sets of two corresponding frequency ranges from the frame spectrum from the other channel to supplement the components reduced by the changes made in the one channel A low-frequency component change stage for changing a ratio between the frequency ranges of the sum of the intensities of the spectrum set ;
For each channel, frequency inverse transformation is performed on the frame spectrum including the modified spectrum set to generate a modified acoustic frame; and
A modified sound frame output step for sequentially outputting the generated modified sound frame for each channel ;
A method for embedding information in an acoustic signal, comprising: An apparatus for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
Sound signal input means for capturing the sound flowing through the reproduction of the acoustic signal with a microphone and digitizing it to obtain an acoustic signal;
Reference frame acquisition means for acquiring, as a reference frame, an acoustic frame composed of a predetermined number of samples from the acquired acoustic signal;
A phase change frame setting means for setting a plurality of acoustic frames set by changing the phase by moving the reference frame and a predetermined sample each as a phase change frame;
Frequency conversion means for performing frequency conversion on the acoustic frame and generating a frame spectrum;
Two sets of spectrum sets of two predetermined frequency ranges are extracted from the generated frame spectrum, and a code determination parameter calculation unit that calculates a sum of the intensities of the spectrum sets for each frequency range as a code determination parameter;
Based on the sum of spectral intensity data calculated in the same in-phase acoustic frames with different reference frames, it is determined that one of the reference frames and the plurality of phase change frames has the optimum phase. And a code output means for outputting a predetermined code based on a ratio between the frequency ranges of the sum value of the spectral intensity data determined for the acoustic frame of the optimum phase,
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: A method for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
An acoustic signal input step of capturing sound that is flowing through reproduction of the acoustic signal with a microphone and digitizing it to obtain an acoustic signal;
A reference frame acquisition step of acquiring an acoustic frame composed of a predetermined number of samples as a reference frame from the acquired acoustic signal;
A phase change frame setting step for setting a plurality of acoustic frames set by changing the phase by moving the reference frame and a predetermined sample each as a phase change frame;
Performing a frequency conversion on the acoustic frame to generate a frame spectrum; and
The spectral set of two predetermined frequency range from the frame spectrum the generated two sets extracted, and the code determination parameter calculating step of calculating a sum of intensities of spectral set as a code decision parameter for each frequency range,
Based on the sum of the intensities of spectrum sets calculated in the same in-phase acoustic frames with different reference frames, the phase of one acoustic frame of the reference frame and the plurality of phase change frames is optimal. A code output stage for determining and outputting a predetermined code based on a ratio between the frequency ranges of the sum of the intensities of the spectrum set determined for the optimal phase acoustic frame;
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 device that reproduces sound while extracting additional information embedded in an inaudible state in advance from an acoustic signal,
A reference frame acquisition means for acquiring, as a reference frame, an acoustic frame composed of a predetermined number of samples from the acoustic signal;
A phase change frame setting means for setting a plurality of acoustic frames set by changing the phase by moving the reference frame and a predetermined sample each as a phase change frame;
Frequency conversion means for performing frequency conversion on the read sound frame and generating a frame spectrum;
Two sets of spectrum sets of two predetermined frequency ranges are extracted from the generated frame spectrum, and a code determination parameter calculation unit that calculates a sum of the intensities of the spectrum sets for each frequency range as a code determination parameter;
Based on the sum of the intensities of spectrum sets calculated in the same in-phase acoustic frames with different reference frames, the phase of one acoustic frame of the reference frame and the plurality of phase change frames is optimal. A code output means for outputting a predetermined code based on a ratio between the frequency ranges of the sum value of the intensities of the spectrum set determined for the acoustic frame of the optimum phase;
Additional information display means for converting the information array corresponding to the outputted code into additional information for display output;
Reproduction frame storage means for storing two or more of the acoustic frames;
Reproduction frame input means for inputting the sound frame read by the sound frame reading means to the reproduction frame storage means;
A new sound frame can be inserted by sound-reproducing the first input sound frame among the sound frames existing in the playback frame storage means and deleting the sound frame from the playback frame storage means after the end of playback. Acoustic frame reproduction means for providing a room for the reproduction frame accumulation means and, when there is a next accumulated acoustic frame, for acoustic reproduction of the next acoustic frame in succession to the first input acoustic frame; ,
An acoustic signal reproducing device comprising: A method of reproducing sound while extracting additional information embedded in an inaudible state in advance from an acoustic signal,
A reference frame acquisition step of acquiring an acoustic frame composed of a predetermined number of samples as a reference frame from the acoustic signal;
A phase change frame setting step for setting a plurality of acoustic frames set by changing the phase by moving the reference frame and a predetermined sample each as a phase change frame;
Performing a frequency conversion on the read sound frame to generate a frame spectrum; and
A code determination parameter calculation step of extracting two sets of spectrum sets of two predetermined frequency ranges from the generated frame spectrum, and calculating a sum value of the intensity of the spectrum set for each frequency range as a code determination parameter;
Based on the sum of the intensities of spectrum sets calculated in the same in-phase acoustic frames with different reference frames, the phase of one acoustic frame of the reference frame and the plurality of phase change frames is optimal. A code output stage for determining and outputting a predetermined code based on a ratio between the frequency ranges of the sum of the intensities of the spectrum set determined for the optimal phase acoustic frame;
An additional information display step of converting the information array corresponding to the outputted code into additional information and displaying it;
A reproduction frame input step of inputting the sound frame read in the sound frame reading step into a reproduction frame storage means for storing two or more sound frames;
It is possible to input a new sound frame by sound-reproducing the first input sound frame among the sound frames existing in the playback frame storage means and deleting the sound frame from the playback frame storage means after the end of playback. An acoustic frame reproduction step of providing a room for the reproduction frame storage means and, when there is a next stored acoustic frame, acoustically reproducing the next acoustic frame in succession to the first input acoustic frame; ,
An acoustic signal reproducing method comprising:

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