JP5104202B2 - Real-time information embedding device for acoustic signals - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a real time embedding device for information to an acoustic signal, reproducing an acoustic signal in which additional information is embedded while the additional information is embedded in the acoustic signal in real time. <P>SOLUTION: When an acoustic signal is input from an audio line input, a recording processing means 210 digitizes the acoustic signal, then creates a recording block formed of a predetermined number of samples and writes the same to a recording block storage means 220. When the recording block is written to the recording block storage means 220, an additional information embedding means 230 reads the written block, embeds additional information to create a reproduced block, and writes the reproduced block to a reproduced block storage means 240. When the reproduced block is written to the reproduced block storage means 240, a reproducing means 250 reads the written block, then converts the same to analog data, and outputs the data as voice to an audio line output. <P>COPYRIGHT: (C)2009,JPO&amp;INPIT

Description

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

Conventionally, as a service for providing music attribute information that allows you to know the title of music that is playing, you can query the broadcast station for the date and time of the broadcasted music, and record music fragments that are played on mobile phones. In addition, there is a service that matches the melody registered in the database. The present applicant has further developed this and proposed a method of embedding attribute information (additional information) by changing the ratio of frequency components of an acoustic signal according to the bit value of the attribute information (Patent Document 1). reference). On the other hand, the present applicant has also proposed a method of extracting additional information in real time while reproducing an acoustic signal in which the additional information is embedded (see Patent Documents 2 and 3).
JP 2006-323246 A JP 2006-195061 A JP 2006-235359 A

  The invention described in Patent Document 1 can embed additional information when there is an acoustic data file in which an acoustic signal is recorded, but can embed additional information when there is no acoustic data file. Can not. Therefore, in live broadcast programs, live performances, lectures, street broadcasts, etc., there is a problem that it is not possible to emit a sound in which additional information is embedded. In addition, there is a problem that even if an acoustic data file exists, it is impossible to meet the demand for changing the additional information while the voice is being emitted.

  Therefore, an object of the present invention is to provide a real-time information embedding device for an acoustic signal capable of reproducing the acoustic signal embedded with the additional information while embedding the additional information in the acoustic signal in real time.

In order to solve the above problems, in the first aspect of the present invention, an acoustic block including a plurality of acoustic frames composed of a predetermined number of samples is sequentially read from the acoustic signal, and each acoustic frame in the acoustic block is read. A recording block that embeds predetermined additional information in an inaudible state and sequentially outputs the acoustic block in which the additional information is embedded as an acoustic signal in which the additional information is embedded, and can store a predetermined number of acoustic blocks A storage means, a reproduction block storage means, a recording processing means for digitizing the acoustic signal to create a single acoustic block consisting of a predetermined number of samples, and writing the acoustic block as a recording block in the recording block storage means; A single recording block stored in the recording block storage means is read and each sound in the recording block is read. Embedding the additional information in the frame, the writes to the reproducing block storage means recording block which the additional information is embedded as a reproduced block, further comprising an additional information setting means, the additional information by the additional information setting means is updated The additional information embedding means for starting embedding from the first bit of the additional information, and the reproduction block stored in the reproduction block accumulating means are converted into analog signals and reproduced as an acoustic signal in which the additional information is embedded. A reproduction processing means, a control for allowing the recording processing means to process without interruption between the recording blocks, and a writing of a single recording block to the recording block accumulating means each time a predetermined recording block is written. Control that causes the additional information embedding means to execute the process once, and if the reproduced acoustic signal is discontinuous. Strangely the have line control to be processed without interruption to the reproduction processing unit, when starting the device, the to start execution of the recording processing means to said reproducing processing means, the additional information embedding means one or more times Provided is a real-time information embedding device for an acoustic signal having execution control means for controlling execution to be started at a delayed time.

  According to the first aspect of the present invention, recording block storage means and reproduction block storage means capable of storing a sound block composed of a predetermined number of samples of a digitized sound signal are prepared and read from the recording block storage means. Additional information is embedded in the extracted recording block and written as a playback block in the playback block storage means, while an acoustic block is created from the input acoustic signal and stored in the recording block storage means. Since the acoustic block is extracted and reproduced as an acoustic signal and control is performed so that these processing timings do not become discontinuous, the additional information can be embedded in the acoustic signal in real time.

In the second aspect of the present invention, an acoustic block including a plurality of acoustic frames composed of a predetermined number of samples is sequentially read with respect to the acoustic signal, and predetermined additional information is added to each acoustic frame in the acoustic block. Is a recording block accumulating means capable of accumulating a predetermined number of acoustic blocks, and a block temporary storage device, which sequentially outputs the acoustic block in which the additional information is embedded as an acoustic signal in which the additional information is embedded. Storage means, reproduction block storage means, recording processing means for digitizing the acoustic signal and reading a single acoustic block consisting of a predetermined number of samples, and writing the acoustic block as a recording block to the recording block storage means, Read a single recording block stored in the recording block storage means, The additional information to each of the sound frames embedded, writes to the block temporary storage means a record block which the additional information is embedded as a reproduced block, further comprising an additional information setting means, the additional information by the additional information setting means Additional information embedding means for starting embedding from the first bit of the additional information when updated, and reading a single playback block stored in the block temporary storage means, and reading the playback block into the playback block storage means A reproduction preparation means for writing to the reproduction block, a reproduction processing means for converting the single reproduction block stored in the reproduction block storage means into an analog signal and reproducing it as an acoustic signal embedded with the additional information, and a gap between the recording blocks. A control for allowing the recording processing means to process the recording processing means so as not to be continuous; and the recording block Each time the writing of a single recording block to the product means is completed, a control for causing the additional information embedding means to execute a predetermined process once, and the reproduction processing means extracts a reproduction block from the reproduction block storage means. each time, a control to execute a single time to the reproduction preparation means a predetermined process, have row control to process without interruption in the reproduction processing means, and that the audio signal reproduced does not become discontinuous, the device And an execution control means for starting execution of the recording processing means at start-up and controlling the reproduction processing means to start execution with a delay when the additional information embedding means is executed at least once. A real-time information embedding device for an acoustic signal is provided.

  According to the second aspect of the present invention, in addition to the first aspect of the present invention, a block temporary storage unit capable of storing an acoustic block is further prepared, and after additional information is embedded, a reproduction block is stored in the block temporary storage unit. Is once written and is further controlled to be written to the reproduction block accumulating means at a predetermined timing, so that it is easier to adjust the timing than in the first mode.

  According to the present invention, it is possible to reproduce an acoustic signal in which the additional information is embedded while embedding the additional information in the acoustic signal in real time.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Usage environment)
First, the use environment of the information real-time embedding device for the acoustic signal according to the present invention will be described. FIG. 1 is a diagram comparing a usage environment of a conventional embedding device and a usage environment of a real-time embedding device of the present invention. Conventionally, as shown in FIG. 1 (a), additional information is embedded in an audio file by an embedding device to obtain an embedded audio file, and the embedded audio file is obtained by a normal playback device. Play and utter sound from the speaker. Additional information is extracted by recording the uttered voice and processing it with an extraction device. In the present invention, as shown in FIG. 1 (b), an acoustic file is reproduced by a normal reproduction device, or sound obtained by a microphone or the like is converted into an acoustic signal by a real-time embedding device, and converted into an acoustic signal. Additional information is embedded, an acoustic signal embedded with the additional information is reproduced, and sound is emitted from a speaker. It is the same as in the past that additional information is extracted by recording this uttered voice and processing it with an extraction device.

  FIG. 2 is a diagram showing the usage environment of the real-time information embedding device for the acoustic signal according to the present invention in more detail. In the present invention, as shown in FIG. 2, the sound obtained by reproducing a sound file with a normal reproducing device, or the sound obtained with a microphone or the like is mixed by a mixer. Then, the mixed acoustic signal is sent to the signal branching unit. The signal splitter separates the mixed audio signal into the channel or track to be embedded and other channels or tracks, and the real-time embedding device embeds the information into the channel or track to be embedded, and the signal delay unit Delays channels and tracks that are not to be embedded in order to match the timing with the embedded signal. Then, the switcher selects either the output signal of the real-time embedding device in which the information is embedded or the output signal of the signal delay device that is not to be embedded based on the operator's instruction and transmits it to the amplifier, and the speaker emits sound. When the additional information is not embedded, the signal branching unit sends the mixed acoustic signal to the signal delay unit and sends it to the amplifier / speaker via the switcher.

(2. First Embodiment)
FIG. 3 is a functional block diagram showing a configuration of a real-time information embedding device for an acoustic signal according to the first embodiment of the present invention. 3, 210 is a recording processing means, 220 is a recording block storage means, 230 is an additional information embedding means, 240 is a reproduction block storage means, and 250 is a reproduction processing means. Note that the apparatus shown in FIG. 3 is compatible with both stereo sound signals and monaural sound signals, but here, a case where processing is performed on stereo sound signals will be described.

  The recording processing means 210 has a recording sound device. The recording processing device 210 has a function of driving a recording sound device and A / D converting an analog signal into digital data, and then outputting a predetermined number of samples as one acoustic block. Have. In this embodiment, the number of samples constituting one sound block is 16384, which is 64 kbytes (0.37 sec in the case of 44.1 kHz, 16 bits, 2ch stereo) for both recording and playback processing. Corresponds to the case. One acoustic block corresponds to a plurality of acoustic frames to be described later. When one acoustic frame is 2048 samples, it corresponds to eight acoustic frames, and when one acoustic frame is 4096 samples, there are four acoustic frames. The recording block storage means 220 has a plurality of buffer memories for storing the sound blocks, and the sound blocks stored 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 recording block accumulating unit 220 has a function of accumulating the sound blocks input from the recording processing unit 210 in the input order and passing them to the additional information embedding unit 230 in that order. Among the sound blocks, those before the additional information is embedded are particularly referred to as recording blocks.

  The additional information embedding unit 230 has a function of embedding additional information by performing processing for each acoustic frame in the acoustic block. Details of the processing in the additional information embedding unit 230 will be described later. Similar to the recording block storage means 220, the reproduction block storage means 240 has a plurality of buffer memories for storing the sound blocks, and has a function of processing the sound blocks stored in these buffer memories by the FIFO method. ing. That is, the reproduction block accumulating unit 240 has a function of accumulating the sound blocks input from the additional information embedding unit 230 in the input order and passing them to the reproduction processing unit 250 in that order. The reproduction processing unit 250 has a reproduction sound device and also has a function of driving the reproduction sound device to reproduce an acoustic block. The reproduction sound device converts the acoustic block which is digital data into D. It has a function of performing A / A conversion and reproducing as sound. Moreover, although not shown, the real-time information embedding device for the acoustic signal of the present invention has execution control means for controlling the execution timing of each means. Of the acoustic blocks, the one after the additional information is embedded is particularly called a reproduction block.

  The apparatus shown in FIG. 3 is actually realized by incorporating a dedicated program into a computer equipped with a sound device for recording and reproduction. The recording block accumulating unit 220 and the reproduction block accumulating unit 240 are realized by securing an area for four acoustic blocks in the memory. The recording processing unit 210 and the reproduction processing unit 250 are realized by programs including a device driver that drives a recording sound device and a reproduction sound device, respectively. The execution control means (not shown) is realized by causing the CPU of the computer to cooperate with the OS and a dedicated program so that the control of each means is executed in parallel by pipeline processing. The computer for realizing the real-time embedding device of the present invention requires two systems of input and output as an audio interface, and preferably has a high CPU capability.

  Next, the processing operation of the apparatus shown in FIG. 3 will be described using the flowchart of FIG. FIG. 4 is a flowchart centering on the processing performed by the execution control means. First, in a state where sound is input from the audio line input, the execution control means activates the recording processing means 210 (S501). The recording processing unit 210 performs a process of A / D converting an input analog signal by a recording sound device and then writing a predetermined number of samples as one acoustic block in the recording block storage unit 220. The recording processing unit 210 transmits a recording end signal to the execution control unit when one writing of the recording block to the recording block storage unit 220 is completed. After starting the recording processing unit 210, the execution control unit waits for a signal from the recording processing unit 210 (S502). In this state, when the recording end signal is received from the recording processing unit 210, the execution control unit causes the additional information embedding unit 230 to read one recording block from the recording block storage unit 220 and also to the additional information embedding unit 230. The recorded block is deleted from the recorded block storage means 220 (S503).

  When the additional information embedding unit 230 reads a recording block from the recording block storage unit 220, the additional information embedding unit 230 executes a process of embedding additional information in the recording block (S504). Details of the additional information embedding process will be described later. When the embedding process is completed for one recording block, the additional information embedding unit 230 transmits an embedding end signal to the execution control unit. Then, the execution control means determines whether or not the reproduction processing means 250 has not been activated (S505). If it has not been activated, the reproduction processing means 250 is activated (S506). When the reproduction processing unit 250 is activated, the additional information embedding unit 230 performs a process of writing the reproduction block that has been embedded in the reproduction block storage unit 240 (S507). The reproduction processing unit 250 sequentially reads the reproduction block written in the reproduction block storage unit 240, performs D / A conversion with the reproduction sound device, and outputs an audio line as an analog signal. If the apparatus stop signal is received in S502, the execution control means performs a process for stopping the apparatus.

(3. Second embodiment)
FIG. 5 is a functional block diagram showing a configuration of a real-time information embedding device for an acoustic signal according to the second embodiment of the present invention. In FIG. 5, components having the same functions as those in FIG. Although not shown, also in the second embodiment, the real-time information embedding device for the acoustic signal has execution control means for controlling the execution timing of each means. In the second embodiment, the first point is that it mainly includes a block temporary storage unit 260 and a playback preparation unit that transfers a playback block from the block temporary storage unit 260 to the playback block storage unit 240. It is different from the embodiment. Note that the reproduction preparation means is not shown.

  Similar to the recording block storage means 220 and the reproduction block storage means 240, the block temporary storage means 260 has a plurality of buffer memories for storing the sound blocks, and the sound blocks stored in these buffer memories are stored in the FIFO method. It has a function to process. That is, the block temporary storage unit 260 has a function of storing the sound blocks input from the additional information embedding unit 230 in the input order and passing them to the reproduction block storage unit 240 in that order. The block temporary storage means 260 is realized by securing an area for four acoustic blocks in the memory.

  Next, the processing operation of the apparatus shown in FIG. 5 will be described using the flowchart of FIG. In FIG. 6, the same processes as those in FIG. The second embodiment is different from the first embodiment in that the process of S508 is performed between the processes of S504 and S505 and that the processes of S509 and S507 are performed when a reproduction end signal is received. .

  In the second embodiment, the execution control means waits for signals from the recording processing means 210 and the reproduction processing means 250 after starting the recording processing means 210 (S501) (S502). In this state, when the execution control means receives the recording end signal from the recording processing means 210, the processing of S503 and S504 is executed.

  When the embedding process is completed for one recording block, the additional information embedding unit 230 writes the reproduction block in the block temporary storage unit 260 (S508). Then, the execution control means determines whether or not the reproduction processing means 250 has not been activated (S505). If it has not been activated, the reproduction processing means 250 is activated (S506). As will be described later, the reproduction processing unit 250 sequentially reads the reproduction blocks written in the reproduction block storage unit 240, performs D / A conversion with the reproduction sound device, and then outputs an audio line as an analog signal.

  In the second embodiment, when a reproduction end signal is received from the reproduction processing unit 250 in the signal waiting state in S502, the reproduction preparation unit reads one reproduction block from the block temporary accumulation unit 260 and stores it in the block temporary accumulation unit 260. The memory area for one acoustic block is released (S509). Then, the reproduction block read from the block temporary storage unit 260 is written into the reproduction block storage unit 240 (S507).

(4. Processing by recording processing means)
Next, the processing by the recording processing unit 210 will be described with reference to the flowchart of FIG. When the recording processing unit 210 is activated by an instruction from the execution control unit (S801), the recording processing unit 210 searches for a free area in the recording block storage unit 220 (S802). If there is a free space in the recording block storage unit 220, the recording processing unit 210 executes the recording process and creates a recording block (S803). When one recording block is created, the recording processing unit 210 writes the created one recording block in the recording block storage unit 220 (S804). When one recording block is written in the recording block storage unit 220, the recording processing unit 210 transmits a recording end signal to the execution control unit (S805).

(5. Processing by reproduction processing means)
Next, the processing by the reproduction processing means 250 will be described using the flowchart of FIG. When the reproduction processing unit 250 is activated by an instruction from the execution control unit (S901), the reproduction processing unit 250 searches for a reproduction block in the reproduction block storage unit 240 (S902). If there is a playback block in the playback block storage unit 240, the playback processing unit 250 reads the playback block from the playback block storage unit 240 (S903). At this time, the reproduction processing unit 250 erases the read reproduction block from the reproduction block storage unit 240 and releases the memory area. Subsequently, the playback processing means 250 executes playback processing for the read playback block (S904). When the reproduction process for one reproduction block is completed, the reproduction processing unit 250 transmits a reproduction end signal to the execution control unit (S905).

(6. Additional information embedding means)
Next, details of the additional information embedding unit 230 will be described. FIG. 9 is a functional block diagram showing details of the additional information embedding unit 230. In FIG. 9, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a frequency component changing means, 40 is a frequency inverse converting means, 50 is a modified acoustic frame output means, 62 is an additional information storage section, and 70 is a bit array. A creation means 80 is a conversion table creation means. Note that the additional information embedding unit 230 shown in FIG. 9 can cope 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 acoustic frame reading means 10 has a function of dividing the recording block read from the recording block accumulating means 220 into a plurality of acoustic frames and reading them in units of acoustic frames. The frequency conversion means 20 has a function of generating a spectrum by frequency-converting the acoustic frame read by the acoustic frame reading means 10 by Fourier transformation or the like. The frequency component changing unit 30 extracts a plurality of spectrum sets corresponding to a predetermined frequency range from the generated spectrum, and based on the bit arrangement created by the bit arrangement creating unit 70 from the additional information extracted from the additional information storage unit 62. The function of changing the state of the spectrum set is provided. The frequency inverse transform means 40 has a function of generating a modified acoustic frame by performing frequency inverse transform on a plurality of spectra including the changed spectrum set. The modified sound frame output unit 50 sequentially stores the generated modified sound frames, and when one sound block is stored, the playback block storage unit 240 (in the case of the first embodiment) or the block temporary storage is stored as an sound block. It has a function of outputting to means 260 (in the case of the second embodiment).

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

  The additional information is information that should be added to the sound information and embedded, and includes attribute information such as a title and artist name, and other information other than the attribute information. In the present embodiment, Nw and Nh are set to Nw = 7 and Nh = 12, respectively. In the present embodiment, since the ASCII code is adopted as the code format of the additional information, Nw = 7 in the additional information, and 7 bits are one word. The bit array created by the bit array creating means 70 is 12 bits, and after the bit array is created, this is processed as one word. Each component shown in FIG. 9 is actually realized by incorporating a dedicated program into a computer. That is, the computer executes the contents of each means according to a dedicated program.

  Next, the processing operation of the additional information embedding unit 230 shown in FIG. 9 will be described. The acoustic frame reading means 10 reads a predetermined number N of samples as one acoustic frame from each of the left and right channels of the recording block read from the recording block storage means 220. The number N of samples of one sound frame read by the sound frame reading means 10 can be set as appropriate, but is desirably about 4096 samples when the sampling frequency is 44.1 kHz. Therefore, the acoustic frame reading means 10 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames.

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

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

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

  In this embodiment, the window function is used, but the window function to be used is divided into the A type acoustic frame and the B type acoustic frame. In this embodiment, the first window function W (1, i) and the second window function W (2, i) as shown in FIGS. 10A and 10B are prepared so that the extraction side can easily recognize them. I made it. The first window function W (1, i) is for use with an A type acoustic frame, and has a maximum value of 1 at the position of a predetermined sample number i as shown in FIG. In the rear part, the minimum value is set to zero. Which sample number takes the maximum value depends on the design of the window function W (1, i), but in this embodiment, it is defined by [Equation 1] described later. The Fourier transform for the A type acoustic frame is performed on the product of the window function W (1, i).

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

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

  In the present embodiment, the window functions W (1, i) and W (2, i) are defined by the following [Equation 1] and [Equation 2]. In FIG. 10, the horizontal axis is the time axis (i). As described later, i is a serial number assigned to N samples in each acoustic frame, and is proportional to time t. 10A and 10B, the vertical axis indicates the amplitude value (level) of the signal. 10A and 10B, the vertical axis indicates the values of the window functions W (1, i) and W (2, i), and the maximum values of W (1, i) and W (2, i). Are all 1.

[Formula 1]
When i ≦ N / 4, W (1, i) = 0.5−0.5 cos (4πi / N)
When N / 4 <i ≦ 11N / 16, W (1, i) = 1.0
When 11N / 16 <i ≦ 13N / 16, W (1, i) = 0.5−0.5 cos (8π (i−9N / 16) / N)
When i> 13N / 16, W (1, i) = 0.0

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

  As is clear from FIG. 10 and [Formula 1] and [Formula 2], the window functions W (1, i) and W (2, i) are asymmetrical to each other. This is for facilitating identification between the two on the extraction side described later.

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

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

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

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

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

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

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

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

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

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

  In the above [Equation 5], m and M are the lower limit and upper limit of the frequency band to be changed, and Zo = M−m. In the present embodiment, Zo = 288.

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

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

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

  In the present embodiment, using the principle of sound pulse segregation, the components of the change target frequency band of the acoustic frame are changed to two states, and 1-bit information is embedded. Here, the state of change of the predetermined frequency component of the acoustic frame before and after the embedding process will be described. FIG. 11 shows the states of the predetermined frequency components of the A type and B type L channel 1 sound frames according to this embodiment. The R channel is omitted because it is the same as the L channel. In each acoustic frame shown in FIG. 11, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction.

  In FIG. 11, the frequency region is divided into six in the frequency direction of the vertical axis, but the second to fifth regions from the top, that is, the frequency band from F1 to F2 is the change target frequency band. The uppermost part, that is, the frequency exceeding F2, and the lowermost part, that is, less than F1, are frequency bands not to be changed. That is, in the present embodiment, the intensity of the spectrum set is changed by setting the frequency F1 to F2 as a predetermined frequency range. As shown in FIG. 11A, regarding the change target frequency band of the A type acoustic frame, the spectrum set is represented by 1D1, 1D2, 1U1, and 1U2 in order of increasing frequency. Also, as shown in FIG. 11 (b), regarding the change target frequency band of the B-type acoustic frame, the spectrum set is represented by 2D1, 2D2, 2U1, and 2U2 in order of increasing frequency.

  When the code 1 is embedded by the 1-bit embedding method, as shown in FIGS. 11C and 11E, the product of the strengths of 1D1 and 2D2 and the product of the strengths of 1U1 and 2U2 are changed to a relatively strong state. The product of the strengths of 1D2 and 2D1 is changed to a relatively weak state. This state is referred to as “state 1”. When embedding the code 2, as shown in FIGS. 11D and 11F, the product of the strength of 1D2 and 2D1 is changed to a relatively strong state, and the product of the strength of 1U2 and 2U1 is changed to a strength of 1D1 and 2D2. The product of the intensity of 1U1 and 2U2 is changed to a relatively weak state. This state is referred to as “state 2”. The shaded portions indicate that the ones having the same concentration are a spectrum set that is a set for obtaining a product. The darker shaded color indicates a group that is changed to a relatively strong state.

  In the 1-bit embedding method, information is embedded by changing the frequency components of the A-type and B-type sound frames into two states as shown in FIGS. 11C, 11E, 11D, and 11F. Is going. Since there are two states, this corresponds to an information amount of 1 bit. As shown in FIGS. 11C to 11F, in the case of the 1-bit embedding method, the upper frequency band and the lower frequency band of the change target frequency band have exactly the same pattern. In the case of the 2-bit embedding method, by independently changing the upper frequency band and the lower frequency band, it is possible to represent 1 bit in the upper frequency band and 1 bit in the lower frequency band.

  In the present embodiment, the change target frequency bands F1 to F2 are set to “0.34 kHz to 3.4 kHz”. This is due to the following reasons. That is, when a mobile phone having a high degree of spread as voice communication is used as an extraction device, the upper limit needs to be 3.4 kHz which is the upper limit of the telephone line band and the mobile phone. This is because the low-pass filter mounted inside the mobile phone corresponds to 3.4 kHz or less in accordance with the telephone exchange. Also, the lower limit of the upper frequency band is set to 1.7 kHz, which is one octave lower than the upper limit of 3.4 kHz. The upper limit of the lower frequency band is set to 1.7 kHz, similar to the lower limit of the upper frequency band, and the lower limit of the lower frequency band needs to be 0.3 kHz or more, which is the lower limit of the telephone line band and the mobile phone. This is because the high-pass filter mounted inside the mobile phone corresponds to 0.3 kHz or more according to the telephone exchange. Therefore, the upper limit of 1.7 kHz is set to 0.34 kHz, which is slightly lower by 2 octaves. The frequency range of the lower frequency band is slightly narrower than the frequency range of the upper frequency band, but the intensity of the distributed signal components is about 4 times on average, so the sensitivity of the lower frequency band is greater than that of the upper frequency band. Can be estimated. The values “0.34 kHz”, “1.7 kHz”, and “3.4 kHz” are representative values, and are not necessarily accurate values, and may be slightly deviated from them.

In the example illustrated in FIG. 11, the description has been given of changing to a relatively strong state and a weak state. However, the level of this strength can be set according to the situation. As will be described below, the larger the ratio between the two, the higher the accuracy at the time of extraction. However, the ratio of interpolation becomes incomplete, and noise due to discontinuous components is heard during reproduction. On the other hand, as the ratio between the two is equal, the reproduction quality approaches the original sound, but the embedded bits cannot be extracted, and the reproduction quality and extraction accuracy are in a trade-off relationship. For example, when the strong side is set to 100% and the weak side is set to 0%, the sound of the portion to be interpolated is 50% of the sound produced by the original acoustic signal before the change as shown in FIG. % Has been confirmed. Therefore, when the strong side is set to 70% and the weak side is set to 30%, the sound of the portion to be interpolated is almost the same as the sound played by the original acoustic signal before the change as shown in FIG. It has been confirmed that this ratio is the limit that can maintain the extraction accuracy. For this reason, it is preferable to set the intensity ratio of the relatively strong spectrum set and the relatively weak spectrum set as 70% and 30%. In order to realize this, in this embodiment, in a specific process described later, a coefficient α = 0.7 for setting a strong state and a coefficient β = 0.3 for setting a weak state are set. However, when the intensity of the spectrum set to be changed to a strong state is originally small, it is necessary to correct the coefficients α and β. For this reason, the frequency component changing means 30 first executes processing according to the following [Equation 6] to thereby obtain the intensities E _1D1 , E _2D1 , E _1D2 , E _2D2 , E _1U1 , E _{2U1 of} each spectrum set. , to calculate the E _1U2, E _2U2.

[Formula 6]
E _1D1 = Σ _{j = m,..., M + G−1} {Al (1, j) ² + Bl (1, j) ² }
E _2D1 = Σ _{j = m,..., M + G−1} {Al (2, j) ² + Bl (2, j) ² }
E _1D2 = Σ _{j = m + G,..., M + 2G−1} {Al (1, j) ² + Bl (1, j) ² }
E _2D2 = Σ _{j = m + G,..., M + 2G−1} {Al (2, j) ² + Bl (2, j) ² }
E _1U1 = Σ _{j = m + 2G,..., M + 2G + Gu-1} {Al (1, j) ² + Bl (1, j) ² }
E _2U1 = Σ _{j = m + 2G,..., M + 2G + Gu-1} {Al (2, j) ² + Bl (2, j) ² }
E _1U2 = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1} {Al (1, j) ² + Bl (1, j) ² }
E _2U2 = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1} {Al (2, j) ² + B1 (2, j) ² }

  In the above [Equation 6], m is the number of the lower limit component of the frequency band to be changed, G is the width of the two divided regions of the lower frequency band in the frequency band to be changed, and Gu is the upper frequency band in the frequency band to be changed. M + 2G + 2Gu is the number of the upper limit component of the frequency band to be changed. For example, when 0.34 kHz to 3.4 kHz is set as the change target frequency band, m = 32 and m + 2G + 2Gu = 320 (= M). Therefore, the width G (= (M / 2−m) / 2) = 64 of the lower frequency band, and the width Gu (= (M−M / 2) / 2) of the upper frequency band. = 80.

  Further, the frequency component changing unit 30 uses the calculated intensity of each spectrum set to calculate the intensity ratio γ of the spectrum set to be changed to a strong state with respect to the spectrum set to be changed to the weak state. This differs depending on whether 1 bit is embedded in one acoustic frame or 2 bits. When 1 bit is embedded in one acoustic frame, the intensity ratio γ is calculated according to the following [Equation 7].

[Formula 7]
If the embedded data is a value _{1, γ = (E 1D1 ·} E 1U1 · E 2D2 · E 2U2) / (E 1D2 · E 1U2 · E 2D1 · E 2U1)
If the embedded data is a value _{2, γ = (E 1D2 ·} E 1U2 · E 2D1 · E 2U1) / (E 1D1 · E 1U1 · E 2D2 · E 2U2)

  When 1 bit is embedded in one acoustic frame, the frequency component changing means 30 further corrects the coefficients α and β by executing the processing according to the following [Equation 8] according to the value of the intensity ratio γ. The coefficients α ′ and β ′ are obtained.

[Formula 8]
In the case of 0.01 ≦ γ <1.0, α'= α · γ -1/4, β'= β · γ 1/4
When γ <0.01, α ′ = 10.0 · α, β ′ = 0.1 · β
When γ ≧ 1.0, no correction is performed.

  On the other hand, when embedding 2 bits in one acoustic frame, the intensity ratio γ is calculated according to the following [Equation 9].

[Formula 9]
When the first embedded data is 1, γ ₁ = (E _1D1 · E _2D2 ) / (E _1D2 · E _2D1 )
When the first embedded data is 2, γ ₁ = (E _1D2 · E _2D1 ) / (E _1D1 · E _2D2 )
If the second embedded data value _{1, γ 2 = (E 1U1} · E 2U2) / (E 1U2 · E 2U1)
If the second embedded data values _{2, γ 2 = (E 1U2} · E 2U1) / (E 1U1 · E 2U2)

When embedding 2 bits in one acoustic frame, the frequency component changing means 30 further executes the processing according to the following [Equation 10] according to the values of the intensity ratios γ ₁ and γ ₂ , thereby obtaining the coefficients α and β. To obtain coefficients α ₁ ′, β ₁ ′, α ₂ ′, β ₂ ′.

[Formula 10]
When 0.01 ≦ γ ₁ <1.0, α ₁ ′ = α · γ ₁ ^−1/2 , β ₁ ′ = β · γ ₁ ^1/2
When γ <0.01, α ₁ ′ = 10.0 · α, β ₁ ′ = 0.1 · β
When γ ≧ 1.0, no correction is performed.
When 0.01 ≦ γ ₂ <1.0, α ₂ ′ = α · γ ₂ ^−1/2 , β ₂ ′ = β · γ ₂ ^1/2
When γ <0.01, α ₂ ′ = 10.0 · α, β ₂ ′ = 0.1 · β
When γ ≧ 1.0, no correction is performed.

  Further, the frequency component changing means 30 is provided with real parts Al (1, j), Ar (1, j), Al (2, j), Ar (2, j) in the continuous A type acoustic frame and B type acoustic frame. , Bl (1, j), Br (1, j), Bl (2, j), Br (2, j) as frequency domain parameters, the lower limit m (= 32) to the upper limit M (= 320) , M to M / 2 are divided into two regions having a width G (= (M / 2−m) / 2), and m + 2G to M are divided into width Gu (= (M−M / 2) / It is divided into two areas having 2), and each is modified according to the bit value to be embedded. As an example, when the bit value to be embedded is “value 1” for both the first bit and the second bit in the method of embedding 2 bits in one acoustic frame, by executing the processing according to the following [Equation 11], The state of the frequency component is changed to “state 1”, that is, the state as shown in FIGS.

[Formula 11]
For each component of j = m to m + G−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α ₁ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α ₁ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α ₁ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α ₁ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β ₁ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β ₁ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β ₁ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β ₁ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + G to m + 2G-1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β ₁ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · β ₁ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β ₁ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β ₁ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α ₁ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α ₁ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α ₁ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α ₁ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + 2G to m + 2G + Gu−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α ₂ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α ₂ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α ₂ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α ₂ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β ₂ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β ₂ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β ₂ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β ₂ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + 2G + Gu to M−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β ₂ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · β ₂ / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β ₂ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β ₂ / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α ₂ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α ₂ / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α ₂ / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α ₂ / {Ar (2, j) ² + Br (2, j) ² } ^1/2

In the method of embedding 2 bits in one acoustic frame, if the bit value to be embedded in either or both of the first bit and the second bit is “value 2”, α ₁ and The processing is executed by exchanging β ₁ with each other or exchanging α ₂ and β ₂ with each other. In the case of a method of embedding 1 bit in one acoustic frame, the processing is executed with α ₁ = α ₂ = α and β ₁ = β ₂ = β in the above [Formula 11]. In this case, the intensity pattern of the spectrum set unit is the same on the high frequency side and the low frequency side.

In addition, when the processing according to the above [Formula 8] and [Formula 10] is executed to obtain the coefficients α ′, β ′, α ₁ ′, β ₁ ′, α ₂ ′, β ₂ ′, 11], coefficients α ′, β ′, α ₁ ′, β ₁ ′, α ₂ ′, β ₂ ′ are used in place of the coefficients α, β, α ₁ , β ₁ , α ₂ , β ₂ .

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

  Specifically, for the A type sound frame, the frequency inverse transform means 40 uses the imaginary part Bl such as the real part Al ′ (1, j) of the left channel of the spectrum obtained by the above [Equation 11]. Using the real part Ar ′ (1, j) of the right channel, imaginary part Br ′ (1, j), etc., according to the following [Equation 12], X ′ '(I) and Xr' (i) are calculated. For the frequency component not modified in the above [Equation 11], Al (1, j) or the like that is the original frequency component is used as Al ′ (1, j) or the like. In calculating the inverse frequency transform, for Al ′ (1, j) and Bl ′ (1, j), Zl (1) in [Formula 5] is replaced with Ar ′ (1, j) and Br ′ (1 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (1) in [Formula 5].

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

In the above [Expression 12], Σ _{j = 0,..., N−1} is shown as Σ _{j in} order to prevent the expression from becoming complicated. The terms “+ Xlp (i + N / 2)” in the first equation and “+ Xrp (i + N / 2)” in the second equation in the above [Equation 12] indicate the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 12], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the A type modified acoustic frame are obtained.

  For the B type acoustic frame, the frequency inverse transform means 40 performs real part Al ′ (2, j) and imaginary part Bl ′ (2, j) of the left channel of the spectrum obtained by the above [Equation 11]. , Using the real part Ar ′ (2, j) and imaginary part Br ′ (2, j) of the right channel, the processing according to the following [Equation 13] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For frequency components that are not modified in the above [Equation 11], in the following [Equation 13], Al ′ (2, j), Bl ′ (2, j), Ar ′ (2, j), Br The original values Al (2, j), Bl (2, j), Ar (2, j), and Br (2, j) are used as ′ (2, j). In calculating the frequency inverse transform, for Al ′ (2, j) and Bl ′ (2, j), Zl (2) in [Equation 5] is replaced with Ar ′ (2, j) and Br ′ (2 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (2) in [Equation 5].

[Formula 13]
Xl' (i + N / 2) = 1 / N · {Σ j Al' (2, j) · cos (2πij / N) / Zl (2) -Σ j Bl' (2, j) · sin (2πij / N ) / Zl (2)} + Xlp (i + N)
Xr' (i + N / 2) = 1 / N · {Σ j Ar' (2, j) · cos (2πij / N) / Zr (2) -Σ j Br' (2, j) · sin (2πij / N ) / Zr (2)} + Xrp (i + N)

  By the above [Equation 13], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified acoustic frame are obtained. The modified sound frame output means 50 sequentially stores the A type modified sound frame and the B type modified sound frame obtained by the processing of the frequency inverse transform means 40, and when one sound block is accumulated, Output.

  Next, the overall flow of processing of the additional information embedding unit 230 shown in FIG. 9 will be described. When the additional information embedding unit 230 is activated, first, the conversion table creation unit 80 creates a code conversion table in which the registration rank of Nw bits and the Hamming code of Nh bits are associated with each other. FIG. 12 shows a flowchart of code conversion table creation by the conversion table creation means 80.

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

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

  The code conversion table created in this way is shown in FIG. As shown in FIG. 13, in the code conversion table, 12-bit codes are registered in association with values 0 to 127 that can be taken by the 7-bit code. As can be seen from the binary representation of the 12-bit code, all the 12-bit codes have a hamming distance of 4 or more. In the example of FIG. 13, for convenience of explanation, a decimal representation of a 7-bit code and a decimal representation and a binary representation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  Next, the process of the embedding device after the code conversion table creation process will be described with reference to the flowchart of FIG. Each component which comprises the additional information embedding means 230 shown in FIG. 9 performs the process according to FIG. 14 in cooperation. FIG. 14 corresponds to processing for one word of additional information. The number of bits Nw of one word can be set to an arbitrary number of bits, but as described above, in this embodiment, it is set to substantially 7 bits of the ASCII code.

  In FIG. 14, first, the bit array creation means 70 creates a corresponding bit array by referring to the conversion table creation means 80 for each word of the additional information extracted from the additional information storage unit 62 (S101). Specifically, first, the additional information storage unit 62 is extracted in units of 1 word (7 bits), referring to the code conversion table shown in FIG. Extract.

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

  Next, the frequency component changing unit 30 performs a process of reading one bit from Nh (= 12) bits held in the register (S102). Subsequently, the acoustic frame reading means 10 reads a predetermined number N of samples as one acoustic frame from each of the left and right channels of the recording block read from the recording block storage means 220, and the frequency conversion means 20 performs frequency conversion. Then, amplitude conversion is performed on the obtained frame spectrum (S103). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 2] is performed using the window function W (1, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (1) and Zr (1) and perform amplitude conversion. Similarly, the acoustic frame reading means 10 reads a predetermined number of samples as B type one acoustic frames from the left and right channels of the recording block read from the recording block storage means 220, and the frequency conversion means 20 performs frequency conversion. Then, amplitude conversion is performed on the obtained frame spectrum (S104). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 3] is performed using the window function W (2, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (2) and Zr (2) and perform amplitude conversion. Subsequently, when the frequency component changing unit 30 changes the state of the frequency component of the A type sound frame and the B type sound frame according to the read bit value, the processing according to [Formula 6] and [Formula 8] is executed. Then, the conversion ratios α and β are determined (S105). Using this determined conversion ratio, the frequency component changing unit 30 executes the processing according to the above [Equation 11] according to the value 1 and the value 2 received from the bit array creating unit 70, and changes the frequency band to be changed. The state of the component is changed to either “state 1” or “state 2” (S106).

  Next, the frequency inverse transforming unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the A type acoustic frame is changed by the process of S106. Perform (S107). This inverse amplitude transform is performed by multiplying the spectrum by the inverse of Zl (1) and Zr (1) calculated by [Equation 5]. This inverse frequency transform is naturally performed by the frequency transforming means 20 in S103. It is necessary to support this method. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the real part Ar ′ (1) of the right channel, such as the real part Al ′ (1, j) of the left channel, the imaginary part Bl ′ (1, j), etc. of the spectrum obtained by the above [Equation 11]. , J) and the like, the imaginary part Br ′ (1, j) and the like are used to perform processing according to the above [Equation 12] to calculate Xl ′ (i) and Xr ′ (i). The modified sound frame output means 50 sequentially stores the obtained modified sound frames and stores one sound block.

  Similarly, the frequency inverse transform unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the B type acoustic frame is changed by the process of S106. Perform (S108). Specifically, the inverse amplitude transform is performed by multiplying the spectrum by the reciprocal of Zl (2) and Zr (2) calculated by [Equation 5], and the actual left channel of the spectrum obtained by [Equation 11] is obtained. Using the part Al ′ (2, j) etc., the imaginary part Bl ′ (2, j) etc., the real part Ar ′ (2, j) of the right channel, the imaginary part Br ′ (2, j) etc. Processing according to [Formula 13] is performed to calculate Xl ′ (i) and Xr ′ (i).

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

  In the present embodiment, the case has been described in which the additional information is converted to 12 bits in the code conversion table by converting the additional information into 7 bits per word, and processing for one word of additional information is performed. As long as there is an agreement with the side, it is possible to record one word of additional information in units of other number of bits.

  Of the modified acoustic signal obtained as described above, for the portion in which the additional information is embedded, the component of the frequency band to be changed has two types of state 1 and state 2 in the case of the 1-bit embedding method, In the case of the 2-bit embedding method, there are only a total of four distributions for each bit. However, since the components other than the component of the frequency band to be changed remain as the original acoustic signals, various distributions are made based on the setting of the producer. Further, as is clear from the processing of [Formula 11], the components of the change target frequency band are changed at the same rate in the left channel and the right channel. Therefore, even at a position equidistant from both speakers, the components in the frequency band to be changed are in a relationship to be amplified without being canceled out, and information can be easily extracted.

  In the real-time information embedding apparatus for an acoustic signal according to the present invention, the content of the additional information can be changed during the embedding. In such a case, when additional information is input by an additional information input unit (not shown), the additional information recorded in the additional information storage unit 62 in the additional information embedding unit 230 is updated. When the additional information is updated, the bit array creation means 70 recognizes this and creates a new bit array based on the updated additional information. Further, the bit array creation means 70 performs a process of overwriting the first 12 bits of the newly created bit array on the 12-bit bit string remaining in the register. Then, the frequency component changing unit 30 starts the process of reading bits from the top of the register. In this way, when the additional information is updated, processing for immediately embedding new additional information is started.

  The details of the additional information extraction unit 230 have been described above. Here, embedding and extraction using amplitude conversion performed in the present embodiment will be described with reference to FIGS. 15 to 18. 15 and 16 show an embedding process and an extraction process when amplitude conversion is not used for comparison, and FIGS. 17 and 18 show an embedding process and an extraction process using amplitude conversion in the present embodiment. In these explanatory diagrams, the signal waveform of the acoustic signal assuming that 3-bit information is embedded is shown for the odd and even three pairs of acoustic frames, and the acoustic frames are shown in a state where they do not overlap each other for convenience of explanation. ing. In each figure, the waveform at the left end shows a normal acoustic signal waveform in the time dimension, the horizontal direction is the time axis, the time advances toward the right, and the vertical direction is intensity (amplitude). The waveform at the center or right end represents the sum of spectral components in a predetermined frequency range after frequency conversion as a signal waveform, and is a signal waveform after virtually passing through a bandpass filter. The vertical direction is similarly strong, and is originally significantly lower than the left end, but here, for convenience of explanation, it is enlarged and expressed to the same extent as the left end. Although the levels of the two types of waveforms are originally different, they are represented at the same level here. Actually, there are high frequency components and low frequency components outside the predetermined frequency range as the data after frequency conversion, but these are not shown on the assumption that they are not modified, and are embedded according to FIG. The two signal waveforms are limited to the two frequency band components for performing the above.

  FIG. 15 (a) shows the original sound signal. The first pair of sound frames has a substantially constant intensity throughout, and the center pair of sound frames has a smaller odd frame intensity and an even frame intensity. The last pair of sound frames has a large odd-numbered frame and a small even-numbered frame. FIG. 15 (b) shows the result of frequency conversion performed on such an acoustic signal, and the component to be embedded is divided into two parts as shown in FIG. ) Are expressed separately as two waveforms (b-1) and (b-2). The waveform shape after frequency conversion differs for each frequency and is not necessarily similar to that shown in FIG. 15A, but here it is assumed that it is the same as before frequency conversion. By embedding 3-bit data [0, 0, 0] based on the method described with reference to FIG. 11 with respect to FIGS. 15 (c-1) and (c-2) are obtained. The first pair can express a pattern as shown in FIG. 11. However, the upper and lower ones of the remaining two pairs are required to construct a magnitude relationship in the opposite direction with respect to the original stage. It turns out that an appropriate pattern has not been constructed. FIG. 15D shows the result of performing frequency inverse transform on the result of such modification. In general, the external shape (referred to as an envelope) of an acoustic signal waveform is determined by low-frequency components having a large energy distribution. In this application, these are included in signal components outside a predetermined frequency range, and are modified in the stage of FIG. 15 (c). Therefore, FIG. 15D after frequency inverse transformation has a shape similar to FIG. 15A before frequency transformation.

  FIG. 16 shows how extraction processing is performed on the result of such embedding. FIGS. 16A and 16B correspond to FIGS. 15D and 15C, respectively. In FIG. 16B, when performing bit determination, since the pattern assumed in FIG. 11 is not formed except for the first pair, it is difficult to determine a correct bit.

  Next, a method in which amplitude conversion is introduced in the present embodiment will be described with reference to FIGS. 17 and 18. FIG. 17 (a) shows the same original acoustic signal as FIG. 15 (a), and FIG. 17 (b) obtained by frequency conversion similarly is the same as FIG. 15 (b). Here, FIG. 17C shows the result of amplitude conversion performed in units of six frames. In the case of FIG. 17, since the amplitude in each frame is flat in FIG. 17A, the whole is flat in FIG. 17C, but actually the inside of each frame fluctuates minutely. Since the intra-frame variation is followed even in the stage of FIG. 17C, it does not normally become completely flat as shown in the figure. (Actually, since FIG. 17C is frequency dimension data, the frequency dimension data is not flat and fluctuates.) Also, the conversion magnification is set for each frame, and the upper and lower two frequency components are set. Since the conversion is performed at the same magnification, there is usually a significant difference between the upper and lower sides (however, the time axis direction is relatively uniform). On the other hand, when 3-bit data [0, 0, 0] is embedded, FIGS. 17D-1 and 17D-2 are obtained. Since FIG. 17C, which is the original stage, has a flat waveform, it can be seen that an ideal pattern as shown in FIG. 11 can be easily constructed in all frames. Next, FIG. 17E shows the result of inverse amplitude conversion performed by multiplying the inverse of the magnification set for each frame. At this stage, a pattern similar to that shown in FIG. 15C is often generated, but any shape can be used. Finally, when frequency inverse transformation is performed, FIG. 17F is obtained, and similarly, the shape is similar to that of FIG. 17A of the original signal waveform.

  FIG. 18 shows how extraction processing is performed on the result of such embedding. 18A, 18B, and 18C correspond to FIGS. 17F, 17E, and 17D, respectively. The waveform shape after frequency conversion in FIG. 18B is basically different from that in FIG. 17B, but the calculated amplitude conversion magnification is a similar value, and the amplitude conversion is performed at substantially the same magnification. 18 (c) is obtained. When bit determination is performed in the stage of FIG. 18C, an ideal pattern as shown in FIG. 11 is formed in all frames, so that correct bits can be determined.

  As described above, in the present embodiment in which amplitude conversion is introduced, since the signal component to be embedded is converted and embedded so as to be flat in the time axis direction as shown in FIG. The probability of making an unnatural change that completely reverses the magnitude relationship between the component intensity on the low frequency side and the low frequency side is reduced, and the extraction accuracy on the extraction side can be increased while maintaining the quality.

(7. Information extraction device from acoustic signals)
Next, an apparatus for extracting information from an acoustic signal will be described. FIG. 19 is a configuration diagram illustrating an apparatus for extracting information from an acoustic signal. In FIG. 19, 100 is an acoustic signal input means, 110 is a reference frame acquisition means, 120 is a phase change frame setting means, 130 is a frequency conversion means, 140 is a code determination parameter calculation means, 150 is a code output means, and 160 is additional information. Extraction means, 170 is an acoustic frame holding means, and 180 is a conversion table creation means.

  The acoustic signal input unit 100 has a function of acquiring and inputting a flowing sound as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. Any microphone can be used, regardless of whether it is monaural omnidirectional or stereo directional, as long as the component in the frequency band to be changed can be detected. Even if it is stereo-directional, only one channel needs to be used. In addition, when information is embedded by the additional information embedding unit 230 shown in FIG. 9, the information is reproduced in stereo, but the sound from both speakers is mixed and input regardless of the sound from either the left or right speaker. There is no restriction on the position of the microphone. Of course, when the acoustic signal in which the additional information is embedded is monaurally reproduced, or when the acoustic signal itself in which the additional information is embedded is monaural, the microphone may be directed to the single speaker to be reproduced. This microphone is not highly accurate, and information can be extracted using a microphone with general accuracy. The reference frame acquisition unit 110 has a function of reading an audio frame composed of a predetermined number of samples as a reference frame from the input digital monaural audio signal (or one channel of a stereo audio signal). As the reference frame, A type and B type are set as in the case of embedding. The phase change frame setting means 120 has a function of setting, as a phase change frame, an acoustic frame whose phase has been changed by moving each of the A type and B type reference frames and predetermined samples.

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

  When performing the Fourier transform on the A type acoustic frame, the frequency converting unit 130 applies the window function W to the signal X (i−N / 2 + pN / 6) (i = 0,..., N−1). (1, i) is used to perform processing according to the following [Equation 14] to obtain the real part A (1, j, p) and imaginary part B (1, j, p) of the converted data. Have. p is a phase number and takes an integer value of 0 to 5.

[Formula 14]
A (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .cos (2πij / N)
B (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .sin (2πij / N)

  When the Fourier transform is performed on the B type acoustic frame, the frequency converting unit 130 applies the window function W (2) to the signal X (i + p · N / 6) (i = 0,..., N−1). , I) is used to perform processing according to the following [Formula 15] to obtain a real part A (2, j, p) and an imaginary part B (2, j, p) of the converted data. ing.

[Formula 15]
A (2, j, p) = Σ _{i = 0,..., N-1} W (2, i) · X (i + p · N / 6) · cos (2πij / N)
B (2, j, p) = Σi _{= 0,..., N-1} W (2, i) .X (i + p.N / 6) .sin (2πij / N)

The frequency conversion unit 130 performs amplitude conversion in the same manner as the frequency conversion unit 20. In performing amplitude conversion, first, conversion magnification is calculated. The conversion magnification is calculated by dividing the set value by the average effective value that is the average of the effective intensity values of the spectrum set in the predetermined frequency range. Although the set value can be determined as appropriate, it is necessary to set the same value as in the case of amplitude conversion at the time of embedding. Therefore, in the present embodiment, the set value Zo needs to be 288 (= M−m). Specifically, the conversion magnifications Z (1, p) and Z (2, p) for the A type acoustic frame and the B type acoustic frame, the A type acoustic frame immediately before these, and the B type acoustic frame. The conversion magnifications Z ₋₁ (1, p) and Z ₋₁ (2, p) are calculated by processing according to the following [Equation 16]. By performing the amplitude conversion, all the sound frames are extracted in a state where the amplitude is converted so that the average effective value of each sound frame becomes a set value regardless of the signal strength in the original state. Will be done. In addition, p is a phase number and takes an integer value of 0-5.

[Formula 16]
Z (1, p) = Zo / [Σ _{j = m,..., M−1} {A (1, j, p) ² + B (1, j, p) ² }] ^1/2
Z (2, p) = Zo / [Σ _{j = m,..., M−1} {A (2, j, p) ² + B (2, j, p) ² }] ^1/2
Z ₋₁ (1, p) = Zo / [Σ _{j = m,..., M−1} {A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ² }] ^1/2
Z ₋₁ (2, p) = Zo / [Σ _{j = m,..., M−1} {A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ² }] ^1/2

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

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

Each component corresponding to the B type acoustic frame one frame before the respective components A (1, j, p) and B (1, j, p) of the A type acoustic frame is represented by A ₋₁ (2, j, p), B ₋₁ (2, j, p), one frame corresponding to each component A (2, j, p), B (2, j, p) of the B type acoustic frame The sound frame before the minute is the A-type sound frame before correction. The code determination parameter calculation unit 140 first calculates the basic intensity values E (1, j, p), E (2, j, p), and reverberation components of each spectrum set before calculating the intensity values of each spectrum set. The basic intensity values E ′ (1, j, p) and E ′ (2, j, p) from which the above is removed are calculated by processing according to the following [Equation 17].

[Formula 17]
E (1, j, p) = A (1, j, p) ² + B (1, j, p) ²
E (2, j, p) = A (2, j, p) ² + B (2, j, p) ²
E ₋₁ (1, j, p) = A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ²
E ₋₁ (2, j, p) = A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ²
E ′ (1, j, p) = E (1, j, p) −q · E ₋₁ (2, j, p)
E ′ (2, j, p) = E (2, j, p) −q · E (1, j, p)

  Of the six formulas in [Formula 17], the fifth formula from the top (second from the bottom) is the B type in which N / 2 samples in the first half overlap when focusing on a certain type A acoustic frame. It is for removing the reverberation component from the sound frame. In addition, the sixth expression from the top (first from the bottom) removes the reverberation component from the A-type sound frame in which the N / 2 samples in the first half overlap when focusing on a certain B-type sound frame. Is for. E ′ (1, j, p) ≧ 0, E ′ (2, j, p) ≧ 0, and as a result of processing according to the fifth and sixth equations from the top of [Equation 17], a negative value is obtained. If this happens, set it to 0.

  In the above [Equation 17], q is a coefficient indicating the magnitude of the reverberation component, but this coefficient q has a value less than 1, and as a result of experiment, when N = 4096, q = 0.06, When N = 2048, q = 0.12. When N = 1024, q = 0.24. When N = 512, q = 0.48 is optimal. Then, the intensity values E1, E2, E3, E4, E5, E6, E7, E8 of each spectrum set when the reverberation component is not removed, and the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 when removed. ′, E6 ′, E7 ′, E8 ′ are calculated from the calculated basic intensity values E (1, j, p), E (2, j, p), E ′ (1, j, p), E ′ (2, j, p) is used to calculate based on the following [Equation 18].

[Formula 18]
E1 (p) = Σ _{j = m,..., M + G-1} E (1, j, p)
E2 (p) = Σ _{j = m, ..., m + G-1} E (2, j, p)
E3 (p) = Σ _{j = m + G,..., M + 2G-1} E (1, j, p)
E4 (p) = Σ _{j = m + G,..., M + 2G-1} E (2, j, p)
E5 (p) = Σ _{j = m + 2G,..., M + 2G + Gu-1} E (1, j, p)
E6 (p) = Σ _{j = m + 2G,..., M + 2G + Gu-1} E (2, j, p)
E7 (p) = Σ _{j = m + 2G + Gu, ..., m + 2G + 2Gu-1} E (1, j, p)
E8 (p) = Σ _{j = m + 2G + Gu, ..., m + 2G + 2Gu-1} E (2, j, p)
E1 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (1, j, p)
E2 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (2, j, p)
E3 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (1, j, p)
E4 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (2, j, p)
E5 ′ (p) = Σ _{j = m + 2G,...,} M _{+ 2G + Gu−1} E ′ (1, j, p)
E6 ′ (p) = Σ _{j = m + 2G,...,} M _{+ 2G + Gu−1} E ′ (2, j, p)
E7 ′ (p) = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1 E ′} (1, j, p)
E8 ′ (p) = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1 E ′} (2, j, p)

  Eventually, the intensity values E1, E2, E3, E4, E5, E6, E7, and E8 of each spectrum set are calculated by [Equation 17] and [Equation 18], and the type corresponding to each spectrum set is determined. Intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, E8 ′ are calculated by subtracting the value obtained by multiplying the intensity of the spectrum set in the immediately preceding acoustic frame by q. become.

  Also, the code determination parameter calculation unit 140 calculates the code determination parameter C using the intensity values E1, E2, E3, E4, E5, E6, E7, and E8 calculated without removing the reverberation component. In the case of a method in which 1 bit is embedded in one acoustic frame, a candidate code B is provisionally determined and a code determination parameter C is calculated by executing processing according to the following [Equation 19].

[Formula 19]
1) In the case of E1 (p) · E5 (p)> E2 (p) · E6 (p) and E4 (p) · E8 (p)> E3 (p) · E7 (p)
C = E1 (p) .E5 (p). {E4 (p) .E8 (p) -E2 (p) .E6 (p) .E3 (p) .E7 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
2) When E2 (p) · E6 (p)> E1 (p) · E5 (p) and E3 (p) · E7 (p)> E4 (p) · E8 (p)
C = {E2 (p) .E6 (p) .E3 (p) .E7 (p) -E1 (p) .E5 (p) .E4 (p) .E8 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
3) In the case of E1 (p), E5 (p), E4 (p), E8 (p)> E2 (p), E6 (p), E3 (p), E7 (p)
C = {E1 (p) .E5 (p) .E4 (p) .E8 (p) -E2 (p) .E6 (p) .E3 (p) .E7 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
4) In cases other than the above 1) to 3) B = 1 and provisional determination,
C = {E2 (p) .E6 (p) .E3 (p) .E7 (p) -E1 (p) .E5 (p) .E4 (p) .E8 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}

  In addition, the code determination parameter calculation unit 140 performs correction code determination using the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, and E8 ′ calculated by removing the reverberation component. The parameter C ′ is calculated. In the case of the method of embedding 1 bit in one acoustic frame, in the above [Equation 19], by replacing E1 (p) to E8 (p) with E1 ′ (p) to E8 ′ (p), respectively, the candidate code B is substituted. The candidate code B ′ is provisionally determined, and the correction code determination parameter C ′ is calculated instead of the code determination parameter C.

  On the other hand, in the case of a method of embedding 2 bits in one acoustic frame, the code determination parameter calculation unit 140 assumes that the candidate codes are B1 and B2 and the code determination parameters are C1 and C2, respectively. 20] is executed, the candidate code B1 is provisionally determined and the code determination parameter C1 is calculated.

[Formula 20]
1) When E1 (p)> E2 (p) and E4 (p)> E3 (p) B1 = 0 and provisional determination
C1 = {E1 (p) · E4 (p) −E2 (p) · E3 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
2) When E2 (p)> E1 (p) and E3 (p)> E4 (p)
C1 = {E2 (p) · E3 (p) −E1 (p) · E4 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
3) In the case of E1 (p) · E4 (p)> E2 (p) · E3 (p) B1 = 0 is temporarily determined,
C1 = {E1 (p) · E4 (p) −E2 (p) · E3 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
4) In cases other than the above 1) to 3) B1 = 1 and provisional determination,
C1 = {E2 (p) · E3 (p) −E1 (p) · E4 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}

  Further, by executing a process according to the following [Equation 21], the candidate code B2 is provisionally determined and the code determination parameter C2 is calculated.

[Formula 21]
1) When E5 (p)> E6 (p) and E8 (p)> E7 (p) B2 = 0 and provisional determination
C2 = {E5 (p) .E8 (p) -E6 (p) .E7 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
2) When E6 (p)> E5 (p) and E7 (p)> E8 (p)
C2 = {E6 (p) .E7 (p) -E5 (p) .E8 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
3) In the case of E5 (p) · E8 (p)> E6 (p) · E7 (p) B2 = 0 is temporarily determined,
C2 = {E5 (p) .E8 (p) -E6 (p) .E7 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
4) In cases other than the above 1) to 3) B2 = 1 and provisional determination,
C2 = {E6 (p) .E7 (p) -E5 (p) .E8 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}

  In addition, the code determination parameter calculation unit 140 performs correction code determination using the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, and E8 ′ calculated by removing the reverberation component. Parameters C1 ′ and C2 ′ are calculated. In the case of the method of embedding 2 bits in one acoustic frame, in the above [Equation 20] and [Equation 21], by replacing E1 (p) to E8 (p) with E1 ′ (p) to E8 ′ (p), respectively, The candidate codes B1 ′ and B2 ′ are provisionally determined instead of the codes B1 and B2, and correction code determination parameters C1 ′ and C2 ′ are calculated instead of the code determination parameters C1 and C2.

  The code output means 150 determines what is determined to be the optimum phase from the acoustic frames (reference frame and phase change frame) corresponding to one reference frame, and selects a code corresponding to the state of the acoustic frame. It has a function to output. The code determination parameter calculation unit 140 and the code output unit 150 constitute an encoding unit. The additional information extraction unit 160 extracts the binary array output by the code output unit 150 in units of Nh bits, refers to the code inverse conversion table, converts it to the registration rank of Nw bits, and further, a predetermined rule It has a function to extract as additional information having meaning after conversion by the above. The acoustic frame holding means 170 is a buffer memory capable of holding two consecutive reference frames (a total of four reference frames for each channel) for each of the A type and B type for each channel. Similarly to the conversion table creation unit 80 shown in FIG. 9, the conversion table creation unit 180 has Nh (having a Hamming distance of at least 4 or more with respect to all the 2 Nw power registration ranks that can be taken by Nw bits. > Nw) It has a function of creating a code conversion table in which a registration order of Nw bits and a Nh-bit Hamming code are associated by assigning a Hamming code of Nw bits. Each component shown in FIG. 19 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, it is desirable to use a portable terminal device such as a cellular phone as hardware. Note that in this specification, even a portable terminal device such as a cellular phone has an arithmetic processing function as a kind of computer.

  Next, the processing operation of the apparatus for extracting information from the acoustic signal shown in FIG. 19 will be described. When the extraction device is activated, first, the conversion table creation means 180 creates a code conversion table. The code conversion table is created by the conversion table creating unit 180 in the same manner as the conversion table creating unit 80, and the code conversion table as shown in FIG. 13 is obtained.

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

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

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 100 as a reference frame (S201). Specifically, reference frames for A type and B type are extracted and read into the acoustic frame holding means 170. The number of samples of one sound frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the sound frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples for each of the A type and B type as reference frames. The acoustic frame holding unit 170 can store two reference frames of A type and B type for each channel, that is, 2.5N samples. When a new reference frame is read, the old reference frame is discarded. It is supposed to be. Therefore, the sound frame holding means 170 always stores four reference frames (continuous 10240 samples).

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

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

  Subsequently, the code determination parameter calculation unit 140 performs a code determination process (S302). Details of the code determination process are shown in FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain each window spectrum (S401). Specifically, the processing according to the above [Equation 14] and [Equation 15] is executed, and the real part A (1, j, p), imaginary part B (1, j, p), real part A of the converted data is executed. (2, j, p) and imaginary part B (2, j, p) are obtained.

  Regarding the processing of S401, in practice, after frequency conversion is performed on the A type sound frame, the frequency conversion is performed on the B type sound frame after being shifted by N / 2 samples. For these conversion data A (2, j, p), B (2, j, p), A (2, j, p), B (2, j, p), the above [Expression 16] is followed. The process is executed to perform amplitude conversion (S402).

  By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, after the code determination parameter calculation unit 140 calculates the code determination parameters C and C ′ as described above, the code determination parameters C and C ′ are used to determine the state of the component of the change target frequency band. In other words, a process of determining what value is embedded as a 1-bit value is performed (S403). Specifically, the code determination parameters C and C ′ are calculated by executing the processing according to the above [Equation 17] to [Equation 19]. Then, comparing these two, if C> C ′, the candidate code B is set in the candidate code table B (p), and if C ≦ C ′, the candidate code B ′ is set in the candidate code table B (p). Output to.

  As a specific processing procedure of S403, first, the basic intensity values E (1, j, p), E (2, j, p) are used by using the first and second expressions of [Expression 17]. ), E1 (p), E2 (p), E3 (p), E4 (p) are calculated using the first to fourth formulas of [Formula 18]. 19], the candidate code B and the code determination parameter C are calculated. Subsequently, reverberation correction processing is performed using the third to sixth expressions of [Expression 17] to obtain basic intensity values E ′ (1, j, p), E ′ (2, j, p). Is calculated. Then, E1 ′ (p), E2 ′ (p), E3 ′ (p), and E4 ′ (p) are calculated using the fifth to eighth formulas of [Formula 18]. 19] to calculate the candidate code B ′ and the correction code determination parameter C ′.

  If 2 bits are embedded in one acoustic frame, in S403, the code determination parameter calculation unit 140 calculates the code determination parameters C1, C2, C1 ′, and C2 ′ as described above, and then determines the code determination parameter. Processing to determine what state of the component of the frequency band to be changed is, that is, what value is embedded as a 2-bit value using C1, C2, C1 ′, and C2 ′ I do. Specifically, processing according to the above [Equation 17], [Equation 18], [Equation 20] and [Equation 21] is executed to calculate the code determination parameters C1, C2, C1 ′ and C2 ′. If C1 + C2> C1 ′ + C2 ′, the candidate codes B1 and B2 are set in the candidate code table B (p), and if C1 + C2 ≦ C1 ′ + C2 ′, the candidate codes B1 ′ and B2 ′ are set in the candidate code table B ( p).

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

[Formula 22]
When C> C ′, S (p) ← S (p) + C
When C ≦ C ′, S (p) ← S (p) + C ′

  If 2 bits are embedded in one acoustic frame, if a value corresponding to 2 bits is output to the candidate code table B (p) at phase p as a result of the determination, in S404, the following [Equation 23 ], The phase determination table S (p) is updated.

[Formula 23]
When C1 + C2> C1 ′ + C2 ′, S (p) ← S (p) + C1 + C2
When C1 + C2 ≦ C1 ′ + C2 ′, S (p) ← S (p) + C1 ′ + C2 ′

  Returning to the flowchart of FIG. 21, the code determination parameter calculation unit 140 stores the code temporarily determined in the code determination process (S302) at the phase p in the candidate code table B (p) (S304).

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

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

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

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

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

  The additional information extraction unit 160 first performs initialization processing (S701). Specifically, a 7-bit code KF and a minimum hamming distance HD are defined, and a process for initializing KF = 0 and HD = 2 is performed. Subsequently, i = 0 is initially set (S702). Next, a 12-bit bit array stored in the buffer is set to a 12-bit code HF, and a Hamming distance hd with a 12-bit code corresponding to the i-th code conversion table is calculated. If hd <HD, KF = i , HD is updated to hd (S703). Then, the value of i is incremented by 1 (S704), and if the value of i is less than 128, the process of S703 is repeated. The additional information extraction means 160 calculates the hamming distance hd for all 128 hamming codes recorded in the code conversion table shown in FIG. 13, and the minimum hamming distance HD and 7 bits when the hamming distance is minimum. A reference code KF is obtained.

  When the minimum hamming distance HD is obtained, the process returns to FIG. 20 to determine whether the obtained minimum hamming distance HD is 0 or 1 or more (S207). If it is determined in S207 that the minimum Hamming distance is 1 or more, the process returns to S201 to perform processing for extracting the next reference frame.

  If it is determined in S207 that the minimum Hamming distance is 0, the additional information extraction unit 160 adds 1 bit to the 7-bit reference code KF obtained by the processing of FIG. 23 and outputs it (S208). If the minimum Hamming distance is 0 in S207, there is a high possibility that the first bit in the buffer is the first bit of the word in the additional information. Therefore, by adding bit “0” to the 7 bits from the head, it is output as one word in the ASCII code. When the Hamming distance is 1 or more, there is a high possibility that the 12 bits used for collation are shifted from the word in the additional information. In this case, the first 1 bit is discarded, and processing for obtaining a new 1 bit is performed by the processing from S201 to S204.

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

  By executing the process shown in FIG. 20 for each reference frame, additional information is extracted. If it is determined in S201 that all reference frames have been extracted, the process ends.

  In the process of S208, the additional information extraction unit 160 first converts the value output by the code output unit 150 according to a predetermined rule and extracts it as meaningful additional information. As the predetermined rule, various rules can be adopted as long as the information intended by the person who embeds the information can be recognized by the recipient, but in this embodiment, the ASCII code is adopted. is doing. That is, the additional information extraction unit 160 recognizes the bit value array obtained from the code determined by the code determination parameter calculation unit 140 and output from the code output unit 150 in units of 1 byte (8 bits), and recognizes this as ASCII. Recognizes character information according to the code. The character information thus obtained is displayed and output on a screen of a display device (not shown).

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

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

  Next, the processing operation of the extraction apparatus in a setting for performing 1-bit error correction will be described with reference to the flowchart of FIG. 24 includes a portion that performs the same processing as in FIG. Therefore, parts that perform the same processing as in FIG. Also in the example of FIG. 24, first, initialization processing is performed (S200). In this initialization process, as in the example of FIG. 20, the phase determination table S (p), the phase determination log, the phase determination flag, and the bit counter are initialized, and further, the automatic correction mode is set to OFF.

  Subsequently, as in FIG. 20, after extracting an acoustic frame composed of a predetermined number of samples as a reference frame (S201), after identifying the phase of each read acoustic frame, the embedded information is determined. The corresponding code is output (S202). Note that the processing of S202 is as shown in FIGS.

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

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

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

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

  If the automatic correction mode is ON, the additional information extraction unit 160 adds 1 bit to the 7-bit reference code KF obtained by the processing of FIG. 23 and outputs it (S208). Then, the bit counter is initialized to 0 (S209).

  As shown in the example of FIG. 24, in the case of setting for 1-bit error correction, it is possible to output 1 word of additional information even if a 1-bit error occurs.

  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. As described with reference to the flowchart of FIG. 21, in the present embodiment, when the same phase continues for a predetermined number of times, the processing is performed after that phase is determined.

(8. When the number of bits in the bit array is Nh and the Hamming distance is changed)
In the above embodiment, when the bit array is created, the created bit array is 12 bits (Nh bits) and the Hamming distance is 4. However, the bit number Nh of the bit array and the Hamming distance are appropriately changed. Is possible. In general, if the bit number Nh of the bit array is increased, the coding efficiency is deteriorated, but the number of error bits that can be corrected is increased. Therefore, when using an extraction device with a relatively high microphone sensitivity and arithmetic processing capability, a bit array with a small number of bits and a hamming distance is created, and an extraction device with a relatively low microphone sensitivity and arithmetic processing capability is used. When used, a bit array having a large number of bits and a hamming distance is created.

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

  The code conversion table created in this way is shown in FIG. As shown in FIG. 25, a 16-bit code is registered in the code conversion table in association with each value 0 to 127 that can be taken by a 7-bit code. As can be seen from the binary representation of the 16-bit code, all the 16-bit codes have a Hamming distance of 6 or more. In the example of FIG. 25, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 16-bit code are shown. Code bit strings are registered in association with each other.

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

  At the time of extraction, the conversion table creation means 180 first creates a code conversion table. After the code conversion table is created, additional information is extracted by the processing according to FIGS. In the case of handling a 16-bit code, in S205 in FIGS. 20 and 24, determination is made based on whether the bit counter is 15 or less or 16 or more. Further, in S701 of FIG. 23, initialization is performed with the minimum hamming distance HD = 3, and in S703, HF is defined as a 16-bit code. Further, the process proceeds to S212 in FIG. 24 when the minimum hamming distance is 3 or more.

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

  In this case, the acoustic frame reading means 10 shown in FIG. 9 has a function of adding or subtracting the same value in units of a predetermined number of samples within a predetermined amplitude range after reading a predetermined number of samples as one acoustic frame. Yes. In this embodiment, with respect to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1), an amplitude range of −32 to +32 (Xl (i), Xr ( i) is a value between −32768 to +32767) and a uniform random number H is generated, and the same value H is used between K (K = 5 in the present embodiment) samples as shown in the following [Equation 24]. The processing as shown in FIG. 5 is executed to update the values of Xl (i) and Xr (i).

[Formula 24]
Xl (i) ← Xl (i) + H
Xr (i) ← Xr (i) -H

  That is, white noise is generated in one acoustic frame by executing the processing according to [Equation 24] over N samples. This white noise generation process is performed immediately after the acoustic frame extraction process of S103 and S104 in the flowchart of FIG.

  As described above, even when white noise is generated, the configuration of the extraction device for extracting information from the acoustic signal on the extraction side is the same as that of FIG. 19, and the processing operation is shown in the flowcharts of FIGS. It is the same as followed.

(10. When creating a sign reverse conversion table during extraction)
In the above embodiment, the extraction apparatus obtains the corresponding 7-bit reference code KF by executing the processing shown in FIG. 23 with Nh (12 or 16) bits stored in the buffer. In this case, however, the processing shown in FIG. 23 is executed in units of one word, so that the processing load increases. In order to reduce such a processing load, it is desirable to create a code reverse conversion table for converting the obtained Nh bits into Nw bits. Here, a case where such a code reverse conversion table is used will be described.

  In this case, the conversion table creation unit 180 of the extraction apparatus has a function of creating a sign reverse conversion table in addition to the function of the conversion table creation unit 80, and the additional information extraction unit 160 performs the process according to FIG. Instead of the function, a function to refer to the code reverse conversion table is provided. Specifically, the additional information extraction unit 160 refers to the code reverse conversion table instead of executing the processing according to FIG. 23 in S206 of FIGS.

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

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

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

  By executing the process shown in FIG. 23 for all of HF = 0 to 4095, a code reverse conversion table as shown in FIG. 26 is created. As shown in FIG. 26, in the code reverse conversion table, the values 0 to 4095 that can be taken by the 12-bit code HF are registered in association with the Hamming distance, and the Hamming distance is 1 or less. Corresponding 7-bit code KF is registered. In the example of FIG. 26, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  When Nh = 16, the process shown in FIG. 23 is executed for all of HF = 0 to 65535, whereby a code reverse conversion table as shown in FIG. 27 is created.

  Then, when performing the process of S206 in FIG. 20, if Nh = 12, the corresponding 12-bit code KF and the Hamming distance are obtained by referring to FIG. 26 with the obtained 12 bits, and Nh When = 16, the corresponding 16-bit code KF and the Hamming distance are obtained with reference to FIG. 27 using the obtained 16 bits.

(11. Modified examples of frequency range)
In the above embodiment, in order to improve the extraction accuracy, a method of expanding the embedding area in the entire telephone band is adopted. However, in order to improve the reproduction quality, the width of the lower change target frequency band may be narrower than that in the above embodiment. Is possible. Specifically, the lower limit F1 of the change target frequency band is set to 0.85 kHz. In this case, the upper frequency band is 1.7 kHz to 3.4 kHz, and the lower frequency band is 0.85 kHz to 1.7 kHz. However, as a result of the experiment, there is no significant difference in quality compared to the method in which the embedded area is set over the entire telephone band, but rather the disadvantage of a decrease in extraction accuracy is greater, and as a result, superiority is seen. There wasn't. In addition, when setting the embedding area in the entire telephone band, the applicant narrows the lower frequency band to 0.85 kHz to 1.7 kHz instead of the method shown in the above embodiment, and further separates the frequency band below it. Experiments were also conducted on a method of installing 0.34 kHz to 0.85 kHz in triplicate, but as a result of comparison, it is more effective to suppress noise by combining the lower frequency bands into one of 0.34 kHz to 1.7 kHz, and the sensitivity is also high. It was found that it was magnified about twice. The bass part has a large signal energy component, so it seems that the sensitivity was significantly improved by a slight enlargement. In the end, the method that divides the frequency range into four frequency regions and makes the upper and lower frequency bands close to each other to form a double sound pulse concentration can suppress the noise most for human hearing. all right.

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

  In the above-described embodiment, the case where the device for extracting information from the acoustic signal is realized by a single mobile terminal device such as a mobile phone has been described as an example. However, the device may be realized in cooperation with other computers. good. Specifically, the portable terminal device and the dedicated computer are connected so as to be capable of wireless communication, and among the components of the acoustic signal input unit 100 to the acoustic frame holding unit 170, those having a large calculation load are processed by the dedicated computer. For example, the portable terminal device includes an acoustic signal input unit 100, a reference frame acquisition unit 110, a phase change frame setting unit 120, an additional information extraction unit 160, and a conversion table creation unit 180. The frequency conversion unit 130, the code determination parameter calculation unit 140, the code output means 150, and the sound frame holding means 170 are provided in a dedicated computer so that necessary information can be communicated between them. This makes it possible to perform high-speed processing even when the processing performance of the portable terminal device is low.

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

  Further, the change of the predetermined frequency component by the frequency component changing means 30 is executed according to FIG. 11 and [Equation 11], but various changes can be used for changing the frequency component according to the bit value to be embedded. For example, it is possible to use a technique as shown in Patent Document 1. In this case, on the extraction side, the extraction is naturally performed by a method corresponding to the embedding method.

It is the figure which compared the utilization environment of the conventional embedding apparatus, and the utilization environment of the real-time embedding apparatus of this invention. It is the figure which showed further the utilization environment of the real-time embedding apparatus of the information with respect to the acoustic signal which concerns on this invention. It is a functional block diagram which shows the structure of the real-time information embedding apparatus with respect to the acoustic signal which concerns on the 1st Embodiment of this invention. It is a flowchart centering on the process which an execution control means performs in 1st Embodiment. It is a functional block diagram which shows the structure of the real-time embedding apparatus with respect to the acoustic signal which concerns on the 2nd Embodiment of this invention. It is a flowchart centering on the process which an execution control means performs in 2nd Embodiment. 4 is a flowchart showing an outline of processing of a recording processing means 210. 4 is a flowchart showing an outline of processing of a reproduction processing means 250. It is a figure which shows the detail of the additional information embedding means 230. FIG. It is a figure which shows the time direction window function used by this embodiment. It is a figure which shows the state of a change of the component of a change object frequency band. It is a flowchart which shows the process outline | summary of code conversion table preparation. It is a figure which shows an example of the code conversion table of 7 bit code and 12 bit code. 10 is a flowchart showing an outline of processing of additional information embedding means 230 shown in FIG. 9. It is a figure for demonstrating the embedding process which does not use amplitude conversion. It is a figure for demonstrating the extraction process which does not use amplitude conversion. It is a figure for demonstrating the embedding process using amplitude conversion. It is a figure for demonstrating the extraction process using amplitude conversion. It is a functional block diagram of the extraction apparatus of the information from an acoustic signal. FIG. 20 is a flowchart showing an outline of processing in the setting shown in FIG. 19 where error correction is not performed. FIG. 21 is a flowchart showing details of phase determination and code output in S202 of FIG. It is a flowchart which shows the detail of the code | symbol determination process of S302 of FIG. It is a flowchart which shows the process for obtaining the corresponding 7-bit code KF and the minimum Hamming distance HD when the 12-bit code HF is specified. FIG. 20 is a flowchart showing an outline of processing in a setting for performing 1-bit error correction in the apparatus shown in FIG. 19. FIG. It is a figure which shows an example of the code conversion table of a 7 bit code and a 16 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 12 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 16 bit code.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Sound frame reading means 20 ... Frequency conversion means 30 ... Frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified sound frame output means 62 ... Additional information storage part 70- .. Bit array creation means 80... Conversion table creation means 100... Acoustic signal input means 110... Reference frame acquisition means 120 .. phase change frame setting means 130. Code determination parameter calculation means 150 ... Code output means 160 ... Additional information extraction means 170 ... Sound frame holding means 180 ... Conversion table creation means 210 ... Recording processing means 220 ... Recording block storage Means 230 ... Additional information embedding means 240 ... Reproduction block storage means 250 ... Reproduction processing means 260 ..Block temporary storage means

Claims (14)

A sound block including a plurality of sound frames composed of a predetermined number of samples is sequentially read into the sound signal, and predetermined additional information is embedded in each sound frame in the sound block in an inaudible state, and the addition is performed. A device that sequentially outputs an acoustic block in which information is embedded as an acoustic signal in which additional information is embedded,
Recording block storage means and playback block storage means capable of storing a predetermined number of sound blocks;
A recording processing unit that digitizes the acoustic signal to create a single acoustic block composed of a predetermined number of samples, and writes the acoustic block as a recording block in the recording block storage unit;
The playback block storage means reads a single recording block stored in the recording block storage means, embeds the additional information in each acoustic frame in the recording block, and uses the recording block in which the additional information is embedded as a playback block. writes to, further comprising an additional information setting means, when said additional information has been updated by the additional information setting means, an additional information embedding means for starting an embedded from the first bit of the additional information,
Reproduction processing means for converting the reproduction block stored in the reproduction block storage means into an analog signal and reproducing it as an acoustic signal in which the additional information is embedded;
Control that causes the recording processing means to process without interruption between the recording blocks, and every time writing of a single recording block to the recording block storage means is completed, the predetermined processing is performed on the additional information. a control to execute a single once embedding means, have row control to process without interruption in the reproduction processing means, and that the audio signal reproduced does not become discontinuous at the time of activation of the device, the execution of the recording processing means Execution control means for starting and controlling the reproduction processing means to start execution with a delay when the additional information embedding means is executed at least once ;
A real-time information embedding device for an acoustic signal. A sound block including a plurality of sound frames composed of a predetermined number of samples is sequentially read into the sound signal, and predetermined additional information is embedded in each sound frame in the sound block in an inaudible state, and the addition is performed. A device that sequentially outputs an acoustic block in which information is embedded as an acoustic signal in which additional information is embedded,
Recording block storage means capable of storing a predetermined number of sound blocks, block temporary storage means, reproduction block storage means,
Recording processing means for digitizing the acoustic signal and reading a single acoustic block consisting of a predetermined number of samples, and writing the acoustic block as a recording block to the recording block storage means;
The block temporary storage means reads a single recording block stored in the recording block storage means, embeds the additional information in each acoustic frame in the recording block, and uses the recording block in which the additional information is embedded as a playback block. writes to, further comprising an additional information setting means, when said additional information has been updated by the additional information setting means, an additional information embedding means for starting an embedded from the first bit of the additional information,
A reproduction preparation unit that reads a single reproduction block stored in the block temporary storage unit and writes the reproduction block to the reproduction block storage unit;
Reproduction processing means for analog-converting a single reproduction block stored in the reproduction block storage means and reproducing it as an acoustic signal in which the additional information is embedded;
Control that causes the recording processing means to process without interruption between the recording blocks, and every time writing of a single recording block to the recording block storage means is completed, the predetermined processing is performed on the additional information. A control for causing the embedding means to execute a single time, and a control for causing the playback preparation means to execute a single process each time the playback processing means extracts a playback block from the playback block storage means, and playback is performed. There line control to process without interruption in the reproduction processing means, and that the audio signal does not become discontinuous at the time of activation of the device, to start execution of the recording processing means, to said reproduction processing means, the additional information Execution control means for controlling to start execution with a delay when the embedding means is executed once or more ;
A real-time information embedding device for an acoustic signal. In any one of claims 1 or claim 2,
The acoustic signal is a stereo signal of two or more channels;
The recording block storage means, block temporary storage means, playback block storage means, recording processing means, additional information embedding means, playback processing means, and execution control means execute processing for each channel. Real-time information embedding device for acoustic signals. In any one of claims 1 to 3,
The additional information embedding means is
Sound frame reading means for sequentially reading sound frames from the recording block;
Of the read acoustic frames, odd-numbered and even-numbered ones are A type and the other is B type, and frequency conversion is performed on the A type acoustic frames using a first window function, and the B type Frequency conversion is performed on the acoustic frame using the second window function to obtain a first window spectrum that is a spectrum corresponding to the first window function and a second window spectrum that is a spectrum corresponding to the second window function. Frequency conversion means;
Two spectrum sets that do not overlap each other in a predetermined frequency range are extracted from the generated window spectra so that the frequency width of the higher frequency spectrum set becomes larger, and each spectrum set is reduced to 1/2. The spectrum set obtained by equally dividing the frequency spectrum into four spectrum sets for each window spectrum and extracting the spectrum set from the first window spectrum in order of increasing frequency is 1D1, 1D2, 1U1, 1U2, and from the second window spectrum. When the extracted spectrum set is 2D1, 2D2, 2U1, and 2U2 in order of frequency, based on the bit arrangement to be embedded, the product of the intensity value of 1D1 and the intensity value of 2D2, the intensity value of 1D2, and 2D1 One of the products with the intensity value of the current value is greater than the other by a predetermined ratio, and at the same time, the intensity value of 1U1 and 2U2 And the product of the degrees value, as one of the product of the intensity values of the intensity values and 2U1 of 1U2 is larger than a predetermined ratio than the other, and the frequency component changing means for changing the intensity of each spectral set,
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the modified spectrum set to generate a modified acoustic frame;
Modified acoustic frame output means for sequentially combining the generated modified acoustic frames and creating the reproduction block;
A real-time information embedding device for an acoustic signal. In claim 4 ,
The frequency component changing means, based on the state of 1 bit in the bit array, a product of an intensity value of 1D1, an intensity value of 1U1, an intensity value of 2D2, and an intensity value of 2U2, and an intensity value of 1D2 and 1U2 The intensity of each spectrum set is changed so that any one of the product of the intensity value of 2D1, the intensity value of 2D1 and the intensity value of 2U1 is larger than the other by a predetermined ratio or more. Real-time information embedding device for signals. In claim 4 ,
The frequency component changing means is one of a product of an intensity value of 1D1 and an intensity value of 2D2, and a product of an intensity value of 1D2 and an intensity value of 2D1, based on the state of one bit in the bit array. Change the intensity of each spectrum set so that is greater than a predetermined ratio over the other, and based on the state of the other 1 bit in the bit array, the product of the intensity value of 1U1 and the intensity value of 2U2; Real-time information of an acoustic signal characterized in that the intensity of each spectrum set is changed so that one of the products of the intensity value of 1U2 and the intensity value of 2U1 is greater than a predetermined ratio than the other. Implantation device. In any one of claims 6 claim 4,
The frequency component changing means includes a product of an intensity value of 1D1 and an intensity value of 2D2, a product of an intensity value of 1D2 and an intensity value of 2D1, a product of an intensity value of 1U1 and an intensity value of 2U2, and 1U2. Of the product of the intensity value of 2 and the intensity value of 2U1, the spectrum set on the side to be changed to be larger is multiplied by the coefficient α, and the spectrum set on the side to be changed to be smaller. Is a device for real-time embedding of information with respect to an acoustic signal, characterized in that the intensity is multiplied by a coefficient β smaller than the coefficient α. In claim 7 ,
The frequency component changing means divides the intensity product of the group that changes the intensity product to be relatively large by the product of the intensity of the group that changes the intensity product to be relatively small. When the value γ is smaller than 1, the coefficient α ′ obtained by dividing the coefficient α by the square root of γ and the coefficient β ′ obtained by multiplying the coefficient β by the square root of γ are used in place of the coefficients α and β. A real-time information embedding device for an acoustic signal. In any one of claims 8 claims 4,
The additional information is composed of Nw bits as one word;
The additional information embedding means is
Conversion table creation means for creating a code conversion table in which Nh (> Nw) bits of Hamming codes with a Hamming distance of at least 4 are assigned to all 2 2 Nw registration orders that Nw bits can take. When,
For each word in the additional information, referring to the code conversion table, a bit array creating means for creating a bit array of Nh bits composed of a corresponding Hamming code;
A real-time information embedding device for an acoustic signal, further comprising: In claim 9 ,
The conversion table creating means creates a code conversion table in which 16-bit Hamming codes having a Hamming distance of 6 or more are assigned to all 128 registration orders that 7 bits can take. Real-time information embedding device for acoustic signals. In any one of claims 10 to claim 4,
The acoustic frame reading means reads the A type acoustic frame and the B type acoustic frame by overlapping a predetermined number of samples,
The real-time information embedding device for an acoustic signal, wherein the modified acoustic frame output means outputs the generated modified acoustic frame by connecting it with a preceding modified acoustic frame. In any one of claims 11 claim 4,
The real-time information embedding device for an acoustic signal, wherein the frequency component changing means sets the predetermined frequency range as 0.34 kHz or more and 3.4 kHz or less. In any one of claims 12 to claim 4,
The acoustic frame reading means generates white noise with extremely low amplitude that is difficult to hear during reproduction with the same number of samples as the acoustic frame, and adds the white noise to the acoustic frame. Real-time information embedding device. A program for causing a computer to function as a real-time information embedding device for an acoustic signal according to any one of claims 1 to 13 .

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