JP2003228388A - Method and device for voice code conversion - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To convert data embedded in a voice code into another code without impairing embedded data. <P>SOLUTION: A voice code converting device which converts a 1st voice code generated by converting an input voice by a 1st voice encoding system into a 2nd voice code by a 2nd voice encoding system is characterized by that when optional data are embedded in the received 1st voice code, a code conversion part converts the 1st voice code into the 2nd voice code and an embedded data extraction part extracts the embedded data from the 1st voice code, and a data embedding part embeds the extracted data in the 2nd voice code and sends them out. <P>COPYRIGHT: (C)2003,JPO

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a speech code conversion method and a speech code conversion apparatus, and more particularly to a speech coding apparatus used in a network such as the Internet or a speech coding apparatus used in an automobile / mobile phone system or the like. The present invention relates to a voice code conversion method and a voice code conversion device for converting a voice code encoded by the method into a voice code of another encoding method.

[0002]

2. Description of the Related Art In recent years, the diversification of mobile phone systems, the explosive increase in subscribers, and voice communication (V
It is considered that the amount of communication between different communication systems will increase more and more due to the spread of oice over IP (VoIP). In voice communication systems such as mobile phones and VoIP, a voice encoding technique for compressing voice is used in order to effectively use a communication line. Mobile phones use different voice coding technologies depending on the country or system.
In CDMA, AMR (Adaptive
The Multi-Rate method is adopted. On the other hand, in VoIP, ITU-T Recommendation G.7 is adopted as a voice coding method.
29A is widely used. In the following, the G.729A encoding method and the decoding method will be described, and the differences between the G.729A and the AMR method will be described.

The G.729A encoding system and decoding system are as follows. -Encoder configuration and operation Fig. 18 is a block diagram of an ITU-T Recommendation G.729A system encoder.
In Fig. 18, a predetermined number of samples per frame (= N)
The input signal (voice signal) X of is input to the LPC analysis unit 1 in frame units. If the sampling rate is 8 kHz and the one frame period is 10 msec, one frame is 80 samples. L
The PC analysis unit 1 regards the human vocal tract as an all-pole filter represented by the following equation H (z) = 1 / [1 + Σαi · z ⁻ⁱ ] (i = 1 to P) (1) The coefficient αi (i = 1, ..., P) of is calculated. Here, P is the filter order. Generally, 10 to 12 as p for telephone band voice
The value of is used. The LPC (linear prediction) analysis unit 1 performs LPC analysis using 80 samples of the input signal, 40 samples of the look-ahead and a total of 240 samples of the past signal 120 samples, and obtains the LPC coefficient.

The parameter converter 2 converts the LPC coefficient into an LSP (line spectrum pair) parameter. Here, the LSP parameter is a parameter in the frequency domain that can be mutually transformed with the LPC coefficient, and since the quantization characteristic is superior to the LPC coefficient, the quantization is performed in the LSP domain. The LSP quantizer 3 quantizes the transformed LSP parameter to generate an LSP code and an LS.
P Find the inverse quantized value. The LSP interpolation unit 4 obtains an LSP interpolation value from the LSP dequantization value obtained in the current frame and the LSP dequantization value obtained in the previous frame. That is, one frame is 5ms
It is divided into the first and second subframes of ec, and LPC
The analysis unit 1 determines the LPC coefficient of the second subframe,
The LPC coefficient of the first subframe is not determined. So LSP
The interpolator 4 predicts the LSP dequantized value of the first subframe by an interpolation operation using the LSP dequantized value obtained in the current frame and the LSP dequantized value obtained in the previous frame.

The parameter inverse transforming unit 5 uses the LSP inverse quantized value and L
Each SP interpolation value is converted into an LPC coefficient and set in the LPC synthesis filter 6. In this case, as the filter coefficient of the LPC synthesis filter 6, the LPC coefficient converted from the LSP interpolation value is used in the first subframe of the frame, and the LPC coefficient converted from the LSP dequantized value is used in the second subframe. . In the following, those with a subscript in 1, such as lspi,
1 in li (n), ... Is the letter L in the alphabet. The LSP parameter lspi (i = 1, ..., p) is quantized by the LSP quantizer 3 by scalar quantization or vector quantization, and then the quantization index (LSP code) is transmitted to the decoder side. It

Next, a sound source and gain search process is performed. The sound source and gain are processed in subframe units. First, the excitation signal is divided into a pitch period component and a noise component, the adaptive codebook 7 storing the past excitation signal sequence is used for the quantization of the pitch period component, and the algebraic codebook is used for the quantization of the noise component. Or noise codebook is used. In the following, a speech coding method using two adaptive codebooks 7 and an algebraic codebook 8 as excitation codebooks will be described.

The adaptive codebook 7 outputs N samples of excitation signals (referred to as periodic signals) sequentially delayed by 1 sample corresponding to indexes 1 to L. N
Is the number of samples in one subframe (N = 40), and the latest (L = 40
+39) Has a buffer that stores the pitch period component of the sample. The index 1 identifies the periodic signal consisting of the 1st to 40th samples, and the index 2 identifies the second signal.
The periodic signal composed of the 41st sample is specified, and the periodic signal composed of the Lth to L + 39th samples is specified by the index L. In the initial state, the content of the adaptive codebook 7 contains signals with all amplitudes of 0. For each subframe, the oldest signal in time is discarded by the subframe length and the excitation signal found in the current subframe is adapted. It operates so as to be stored in the codebook 7.

In the adaptive codebook search, the adaptive codebook 7 storing past excitation signals is used to identify the periodic component of the excitation signal. That is, the past excitation signal in the adaptive codebook 7 is extracted by the subframe length (= 40 samples) while changing the starting point read from the adaptive codebook 7 by one sample, and L
It is input to the PC synthesis filter 6 to create a pitch synthesis signal βAPL. However, PL is the past pitch periodic signal (adaptive code vector) corresponding to the delay L extracted from the adaptive codebook 7, A is the impulse response of the LPC synthesis filter 6,
β is an adaptive codebook gain.

The calculation unit 9 obtains the error power EL between the input voice X and βAPL by the following equation EL = │X-βAPL│2 (2). A is the weighted composite output of the adaptive codebook output.
Let PL be the autocorrelation of APL be Rpp, and the cross-correlation of APL and input signal X be Rxp, the adaptive code vector PL at pitch lag Lopt at which the error power in equation (2) is minimized.
Is expressed by the following equation P _L = argmax (Rxp ² / Rpp) (3). That is, the optimum starting point is the read start point at which the value obtained by normalizing the cross-correlation Rxp between the pitch synthesized signal APL and the input signal X by the autocorrelation Rpp of the pitch synthesized signal is the largest. From the above, the error power evaluation unit 10
Find the pitch lag Lopt that satisfies the equation (3). At this time,
The optimum pitch gain βopt is given by the following equation βopt = Rxp / Rpp (4).

Next, the algebraic codebook 8 is used to quantize the noise component contained in the excitation signal. The algebraic codebook 8 has an amplitude of 1
Or, it is composed of a plurality of pulses of -1. As an example, Table 1 shows the pulse positions when the subframe length is 40 samples.

【table 1】 The algebraic codebook 8 divides N (= 40) sample points constituting one subframe into a plurality of pulse system groups 1 to 4,
For all combinations obtained by extracting one sample point from each pulse system group, a pulse signal having +1 or -1 pulse at each sample point is sequentially output as a noise component. In this example, basically four pulses are arranged per subframe.

FIG. 19 is an explanatory view of the sampling points assigned to the pulse system groups 1 to 4, and (1) the pulse system group 1 has eight sampling points 0, 5, 10, 15, 20, 25, 30,3
5 are assigned, and (2) pulse system group 2 is assigned 8 sample points 1,6,11,16,21,26,31,36,
(3) Eight sample points 2,7,1 in pulse system group 3
2,17,22,27,32,37 are assigned, and (4) 16 groups of sampling points are assigned to the pulse system group 4, 3,4,8,9,13,14,18,19,23,
24,28,29,33,34,38,39 are assigned.

3 bits are required to express the sampling points of the pulse system groups 1 to 3, 1 bit is required to express the positive / negative of the pulse, and a total of 4 bits are required, and the sampling points of the pulse system group 4 are expressed. To achieve this, 4 bits are required, and 1 bit is required to express the positive / negative of the pulse, for a total of 5 bits. Therefore, the random codebook 8 having the pulse arrangement in Table 1 is
17 bits are required to specify the pulsed signal output from, and the type of pulsed signal is 217 (= 24 × 24
X24x25) present. As shown in Table 1, the pulse positions of each pulse system are limited, and in the algebraic codebook search, among the combinations of pulse positions of each pulse system, the pulse with the smallest error power with the input voice in the reproduction area is selected. Determine the combination. That is, the optimum pitch gain βopt obtained by the adaptive codebook search is set, and the adaptive codebook output PL
Is multiplied by the gain βopt and input to the adder 11. Simultaneously with this, a pulsed signal from the algebraic codebook 8 is sequentially input to the adder 11, and the output of the adder is input to the LPC synthesis filter 6 to obtain a pulsed signal having a minimum difference between the reproduced signal and the input signal X. Identify the signal. Specifically, first, the optimum adaptive codebook output P obtained by the adaptive codebook search from the input signal X
From L and the optimum pitch gain β _opt, a target vector X ′ for algebraic codebook search is generated by the following equation.

X '= X-β _opt APL (5) In this example, the pulse position and amplitude (positive / negative) are as described above.
Since it is expressed in 17 bits, there are 2 to the 17th power combinations. Here, assuming that the kth algebraic code output vector is Ck, in the algebraic codebook search, a code vector Ck that minimizes the evaluation function error power D of the following equation D = | X'-GCACk | 2 (6) is obtained. . GC is the algebraic codebook gain. The error power evaluation unit 10 searches the algebraic codebook for the algebraic composite signal AC.
The square of the cross-correlation value Rcx of k and the input signal X ′ is normalized by the auto-correlation value Rcc of the algebraic composite signal, and the normalized cross-correlation value (Rcx * Rcx / Rcc) is maximized. Search for combinations.

Next, the gain quantization will be described. G.72
In the 9A method, the algebraic codebook gain is not directly quantized,
Adaptive codebook gain Ga (= βopt) and algebraic codebook gain G
Vector-quantize the correction coefficient γ of c. Here, between the algebraic codebook gain GC and the correction coefficient γ, GC = g ′ ×
There is a relation called γ. g'is the gain of the current frame predicted from the logarithmic gain of the past 4 subframes. In the gain quantization table (not shown) of the gain quantizer 12, 128 combinations (= 27) of the adaptive codebook gain Ga and the correction coefficient γ with respect to the algebraic codebook gain are prepared.
The gain codebook search method is such that, for the adaptive codebook output vector and the algebraic codebook output vector, one set of table values is extracted from the gain quantization table and set in the gain variable units 13 and 14, The respective vectors are multiplied by gains Ga and Gc in the units 13 and 14 and input to the LPC synthesis filter 6, and the error power evaluation unit 10 selects a combination having the smallest error power with the input signal X.

From the above, the line coding unit 15 uses the LSP code which is the quantization index of the LSP, the pitch lag code Lopt which is the quantization index of the pitch lag, (3) the algebraic code which is the algebraic codebook index, and (4) the gain. The gain code, which is the quantization index of, is multiplexed to create line data and transmitted to the decoder.

Configuration and Operation of Decoder FIG. 20 is a block diagram of a G.729A system decoder. The line data sent from the encoder side is input to the line decoding unit 21, and the LSP code, pitch lag code, algebraic code, and gain code are output. The decoder decodes the audio data based on these codes. Regarding the operation of the decoder, it partially overlaps because the function of the decoder is included in the encoder,
A brief description will be given below. When the LSP code is input, the LSP dequantization unit 22 dequantizes the LSP code and outputs the LSP dequantized value. LSP
The interpolator 23 determines the LS in the second subframe of the current frame.
The LSP dequantized value of the first subframe of the current frame is interpolated from the P dequantized value and the LSP dequantized value of the second subframe of the previous frame. Next, the parameter inverse conversion unit 24
The LSP interpolated value and the LSP dequantized value are respectively converted into LPC synthesis filter coefficients. G.729A LPC synthesis filter 25
Uses the LPC coefficient converted from the LSP interpolation value in the first first subframe, and uses the LPC coefficient converted from the LSP dequantized value in the second second subframe.

The adaptive codebook 26 has a subframe length (= 40 samples) from the read start position indicated by the pitch lag code.
, The noise codebook 27 outputs the pulse position and the polarity of the pulse from the read position corresponding to the algebraic code. Further, the gain dequantization unit 28 calculates an adaptive codebook gain dequantization value and an algebraic codebook gain dequantization value from the input gain code and sets them in the gain variable units 29 and 30. The adding unit 31 adds a signal obtained by multiplying the adaptive codebook output by the adaptive codebook gain dequantization value and a signal obtained by multiplying the algebraic codebook output by the algebraic codebook gain dequantization value Create a signal and use this source signal as an LPC
Input to the synthesis filter 25. Thereby, the reproduced voice can be obtained from the LPC synthesis filter 25. In the initial state, the contents of the adaptive codebook 26 on the decoder side are all amplitude 0.
, The oldest signal in time is discarded for each subframe by the subframe length, and the excitation signal obtained in the current subframe is stored in the adaptive codebook 26. That is, the adaptive codebooks 26 of the encoder and the decoder are always maintained in the latest state. That's G.
729A encoding and decoding method. On the other hand, the AMR method is also G.72.
CELP (Code Excited Linear Prediction;
A basic algorithm called code-driven linear predictive coding) is used, and the difference from the G.729A system is as follows.

-Difference in encoding method between G729A system and AMR system Fig. 21 shows a result of comparison of main specifications of the G.729A system and AMR system. Note that there are eight types of AMR coding modes in all, but the specifications in FIG. 21 are common to all coding modes. G729
A method and AMR method, the sampling frequency of the input signal (= 8KHz),
The subframe length (= 5 msec) and the linear prediction order (= 10th order) are the same, but the frame length is different and the number of subframes per frame is different. As shown in FIG. 22, G.729A
In the method, one frame is composed of two 0th to 1st subframes, and in the AMR method, one frame is four 0th to 3rd subframes.
It is composed of subframes.

FIG. 23 shows the result of bit allocation comparison between the G.729A system and the AMR system, and the AMR system shows the case of 7.95 kbit / s mode which is the closest to the bit rate of G.729A. As is apparent from FIG. 23, the number of bits (= 17 bits) of the algebraic codebook per subframe is the same, but the allocation of the number of bits required for other codes is different. Further, in the G.729A system, the adaptive codebook gain and the algebraic codebook gain are collectively vector-quantized, so that there is only one type of gain code per subframe. Two types of algebraic codebook gain are required. As described above, the G.729A method, which is widely used in VoIP for communicating voice over the Internet, and the AMR method, which is adopted in the mobile phone system, have the same basic algorithm, but the frame length is different, and The number of bits expressing the code is different.

Voice code conversion With the spread of the Internet and mobile phones, it is considered that the communication volume of voice calls between Internet users and mobile phone network users will increase more and more in the future. For voice communication between such different communication systems, a voice code conversion device 53 is required in the middle as shown in FIG. That is, in the speech transcoding device 53, the speech code encoded by the encoder 52 of the one communication system 51 according to the first speech coding system is converted into the speech code of the second speech coding system used in the other communication system 54. Convert to voice code. By performing voice code conversion in this way, the decoder 55 of the second voice coding system of the communication system 54 can correctly reproduce the voice of the user 1.

As such a code conversion technique, a tandem connection system in which decoding / coding is repeated by a voice encoding system of each system, or a voice code is decomposed into each element code constituting the voice code, and each element code is A method of individually converting to a code of another voice encoding method has been proposed (Japanese Patent Application No. 2001).
-75427). FIG. 25 is an explanatory diagram of the latter method. The encoder 71a of the encoding system 1 incorporated in the terminal 71 encodes the voice signal of the user A into a voice code of the encoding system 1 and sends it to the transmission line 71b. The voice code conversion unit 74 is a transmission line
The audio code of the encoding method 1 input from 71b is converted into the audio code of the encoding method 2 and transmitted to the transmission line 72b, and the decoder 72a of the terminal 72 inputs the encoding method 2 via the transmission line 72b. The reproduced voice is decoded from the voice code of, and the user B can hear the reproduced voice.

The coding method 1 is L obtained from the linear prediction coefficient (LPC coefficient) obtained by the linear prediction analysis for each frame.
The first L obtained by quantizing the SP parameter
An SP code and a first pitch-lag code that specifies an output signal of an adaptive codebook for outputting a periodic excitation signal,
A first algebraic code (noise code) for specifying an output signal of an algebraic codebook (or a noise codebook) for outputting a noisy excitation signal, a pitch gain representing the amplitude of the output signal of the adaptive codebook, and This is a method of encoding a voice signal with a first gain code obtained by quantizing an algebraic codebook gain representing the amplitude of an output signal of an algebraic codebook. The coding method 2 is a second LSP code, a second pitch lag code, a second algebraic code (noise code), a second LSP code obtained by quantizing by a quantization method different from the first speech coding method.
This is a method of encoding a voice signal with the gain code of.

The voice code conversion section 74 includes a code separation section 74a and LS.
P code converter 74b, pitch lag code converter 74c, algebraic code converter 74d, gain code converter 74e, code multiplexer 74
have f. The code separation unit 74a includes the encoder 71 of the terminal 1.
A code of a plurality of components necessary for reproducing a voice signal from a voice code of the encoding method 1 input from a through the transmission path 71b, that is, an LSP code, a pitch lag code,
It is separated into an algebraic code and a gain code and input to each of the code conversion units 74b to 74e. Each of the code conversion units 74b to 74e receives the LSP code, the pitch lag code, the algebraic code, and the gain code according to the speech coding method 1 from the LSP code, the pitch lag code, the algebraic code, and the gain code (pitch gain with the speech coding method 2). Code, algebraic gain code),
The code multiplexing unit 74f multiplexes the converted codes of the voice coding method 2 and sends them to the transmission line 72b.

Data embedding technology With the spread of computers and the Internet in recent years, multimedia contents (still images, moving images, audio,
"Digital watermarking technology", which embeds special data in (such as voice), is drawing attention. The digital watermarking technique is a technique for embedding other arbitrary information into multimedia contents such as images, moving images, and sounds, by utilizing the characteristics of human perception and hardly affecting the quality. Such technologies often aim to protect copyright by embedding the names of creators and sellers in the content to prevent illegal copying and data tampering. It is also used for the purpose of embedding or additional information to improve the convenience of the user when using the content.

Also in the field of voice communication, attempts are being made to embed such arbitrary information in a voice code for transmission. FIG. 26 is a conceptual diagram of a voice communication system to which the data embedding technique is applied. The encoder 81, when encoding the input voice SP into a voice code, outputs an arbitrary data sequence DT other than voice.
Embedded in the voice code SCD and transmitted to the decoder 82. At this time, since the data is embedded in the voice code itself without changing the voice code format, the information amount of the voice code does not increase. The decoder 82 reads out an arbitrary data sequence embedded in the voice code, performs the normal decoder processing on the voice code, and outputs the reproduced voice SP '. At this time, since the embedding is performed so that the quality of the reproduced sound SP ′ is hardly affected, the reproduced sound is almost the same as the case where the embedding is not performed. With the above configuration, it is possible to transmit arbitrary data separately from voice without increasing the transmission amount. In addition, a third party who does not know that the data is embedded is recognized as normal voice communication.

There are various methods for embedding data. In particular, in the high compression speech coding method based on the CELP method, some methods of embedding arbitrary information in the coded speech code have been proposed. For example, in a speech coding method that performs coding using an algebraic codebook and an adaptive codebook, a technique has been proposed in which arbitrary data is embedded in a pitch lag code or an algebraic code. This embedding technique embeds an arbitrary data sequence in a code (pitch lag code, algebraic code) quantized by an algebraic codebook or an adaptive codebook according to a certain rule. Focusing on the pitch lag code corresponding to the pitch sound source and the algebraic code corresponding to the noise sound source, these gains (pitch gain, algebraic codebook gain) can be regarded as factors indicating the contribution of each code, and when the gain is small Has a smaller contribution of the corresponding code. Therefore, the gain is defined as a determination parameter, and when the gain is less than or equal to a certain threshold value, it is determined that the contribution of the corresponding code is small, and the index of the code is replaced with an arbitrary data sequence. This makes it possible to embed arbitrary data while suppressing the effect of replacement.

In the future, it is expected that communication between communication systems to which the data embedding technique described above is applied will increase. At this time, the voice code conversion device needs to perform the code conversion for the voice code embedded with the data.

[0028]

[Problems to be Solved by the Invention] -Problem 1 FIG. 27 shows a principle diagram of code conversion. FIG. 27 shows a case where coded data Code1 of the first coding method is converted into coded data Code2 of the second coding method. Code converter 91
Includes a first quantization table 92 used for encoding by the first encoding method and a second quantization table 93 used for encoding by the second encoding method.
Further, although the first quantization table 92 and the second quantization table 93 have different table sizes and table values, FIG. 27 shows the case where the table size is the same as 2 bits for simplification of description.

In FIG. 27, the coded data Code1 (“01” in the figure) of the first coding method input to the code conversion unit 91.
Represents the index number of the first quantization table 92. Therefore, the value having the smallest error in the value (2.0 in the figure) of the first quantization table 92 corresponding to the input Code1 is selected from the second quantization table 93, and the corresponding second value is selected.
Index number of quantization table 93 ("10" in the figure)
Is output as encoded data Code2 of the second encoding method.
In this way, the code conversion unit 91 compares the quantization tables of the conversion source and the conversion destination and associates the index numbers so that the error becomes the smallest. Where input code Co
Consider a case where the data series of de1 is arbitrary data ("01") embedded by the above-described embedding method. The code conversion unit 91 converts the input data series “01” into “10” in order to perform the same conversion processing as described above. However, with this, the embedded data sequence changes from "01" to "10" and is no longer held, and the decoder of the second encoding method on the receiving side can restore the embedded data sequence normally. I can't. As described above, in the conventional code conversion method,
When an arbitrary data series is embedded in the input code,
There is a problem that the embedded data series cannot be held and, as a result, the embedded data is lost in the code conversion device.

[Problem 2] As will be represented by the third-generation mobile phone system in the future,
In addition to voice communication, communication systems for multimedia information such as data communication are expected to spread. For this reason,
Communication occurs between a conventional communication system having only voice lines and a communication system having voice lines and other data lines. In such a case, for a voice line, voice communication between users can be performed by performing mutual conversion of voice codes between both communication systems using a conventional voice code conversion device.
However, one of the data lines does not have a data line, so that data communication between users is impossible. As described above, between the communication system having only the voice line and the communication system having the voice line and the other data line, there is a problem that only the voice communication can be performed between the users.

From the above, the first object of the present invention is to embed a voice code encoded by the first encoding method without damaging the embedded data embedded in the voice code. In addition, it is possible to convert to a voice code of the second encoding method. A second object of the present invention is to enable both voice communication and data communication between a communication system having only a voice line and a communication system having a data line outside the voice line.

[0032]

According to a first aspect of the present invention, a first voice code obtained by encoding input voice by a first voice encoding method is used.
A voice code conversion method and device for converting into a second voice code by a two-voice encoding method. In the first voice code conversion device of the present invention, the code conversion unit converts the first voice code to the second voice code when any data is embedded in the received first voice code and embeds it. The data extraction unit extracts embedded data from the first voice code, and the data embedding unit embeds the extracted data in the second voice code obtained by the conversion and sends it out. As described above, the embedded data is once extracted from the voice code of the first encoding method of the conversion source, and the data is embedded again in the voice code of the second encoding method after the code conversion. The voice code can be converted into the voice code of the second encoding method in which the data is embedded without damaging the data embedded in the voice code.

When data is embedded in the first voice code by replacing a part of the first voice code with the data when the data embedding condition is satisfied at the transmission source, the embedded data extracting section transmits the data. Monitor whether or not the data embedding condition is satisfied by referring to the dequantized value of a predetermined element code that constitutes the first speech code received from the original, and if the data embedding condition is satisfied, then from the first speech code The embedded data is extracted, and the extracted embedded data is stored in the data holding unit. The data embedding unit refers to the dequantized value of a predetermined element code constituting the second speech code converted from the first speech code, monitors whether the data embedding condition is satisfied, and if it is satisfied, The data is embedded in the second voice code by replacing a part of the second voice code with the data stored in the data holding unit. In this way, when the embedding control is adaptively performed at the conversion source and the conversion destination, the timing of data extraction and embedding caused by the difference in the embedding control method of each encoding method or by the conversion error in the conventional speech code conversion unit. By absorbing the difference of the above by the data holding unit, it is possible to convert the data embedded in the voice code of the first encoding system into the voice code of the second encoding system in which the same data is embedded.

In the second voice code conversion apparatus of the present invention, the receiving unit receives the first voice code and the data separately from the transmission source, and the code converting unit converts the first voice code into the second voice code, The data embedding unit embeds the data in the second voice code obtained by the conversion and transmits it to the destination. With this configuration, when the voice code and the data of the first coding method are separately input to the voice code conversion unit from the communication system of the conversion source through separate lines or multiple lines, the voice code conversion unit performs the code conversion after the code conversion. By embedding data in the voice code of the second encoding method, it becomes possible to transmit to the conversion destination only by the voice line.

In the third voice code conversion apparatus of the present invention, the code conversion unit converts the first voice code into the second voice code when arbitrary data is embedded in the received first voice code. The embedded data extraction unit extracts embedded data from the first voice code, and the transmission unit transmits the second voice code obtained by the conversion and the extracted data separately to the destination through separate lines or multiplex lines. . With this configuration, the voice information and the data information transmitted by the conversion source voice line can be separately transmitted to the conversion destination voice line and the data line.

[0036]

BEST MODE FOR CARRYING OUT THE INVENTION (A) Outline of the present invention (a) First system FIG. 1 is a conceptual diagram of a first system of the present invention, in which a voice of a first encoding system in which arbitrary data DT is embedded. Code SP1
The case where the data DT is converted into the voice code SP2 of the second encoding method is shown. A voice code conversion device 103 is provided between a communication system 101 of the first coding system and a communication system 102 of the second coding system. When encoding the input voice SP1, the encoder 104 of the first encoding system in the communication system 101 embeds an arbitrary data sequence DT other than voice data in the voice code SCD1 and sends it to the transmission line 105. At this time, since the data is embedded by the encoder 104 in the voice code itself without changing the format of the voice code,
There is no increase in the amount of voice code information.

The voice transcoding device 103 is connected to the encoder 104 from the encoder 104.
When the voice code SCD1 encoded according to the one voice encoding system is received, the voice code is converted into the voice code SCD2 of the second voice encoding system used in the communication system 102 and transmitted to the transmission line 106. At this time, the voice code conversion device 103 performs voice code conversion without damaging the embedded data. In the communication system 102, the decoder 107 of the second coding method is the speech code S
An arbitrary data sequence DT embedded in CD2 is read and output, and a voice code is subjected to a normal decoder process to output reproduced voice SP2. At this time, since the embedding is performed so that the quality of the reproduced sound SP2 is hardly affected, the reproduced sound is almost the same as the case where the embedding is not performed.

FIG. 2 is a block diagram of the code conversion device 103 in the first system of the present invention. The voice code SCD1 encoded at the conversion source according to the first encoding method and having the data DT embedded therein is input to the code conversion unit 111 and the embedded data extraction unit 112 in order on a frame-by-frame basis. The code conversion unit 111 is shown in FIG.
The speech code SCD1 of the first coding method is converted into the speech code SCD2 'of the second coding method, which has the same configuration as the conventional one shown in FIG. The embedded data extraction unit 112 extracts the data DT embedded in the voice code SCD1 and outputs it to the data embedding unit 113. The data extraction method by the embedded data extraction unit 112 is the same as the data extraction method of the decoder of the first encoding method. The data embedding unit 113 includes a voice code SCD2 ′ and a voice code SCD of the second coding method converted by the code conversion unit 111.
When the data DT extracted from 1 is input, the voice code SCD2 ′
Data DT is embedded in each frame to
Output as SCD2. The data embedding method by the data embedding unit 113 is the same as the data embedding method of the encoder of the second coding method.

FIG. 3 is another block diagram of the code conversion apparatus 103 in the first system of the present invention, in which the same parts as those of the code conversion apparatus of FIG. This code conversion device 103 adaptively adjusts the voice code SCD based on the nature of the voice code.
The embedded data DT is extracted from 1 and the voice code SCD
Embed the data DT in 2 '. For example, as described in the section of the prior art, the encoder of the first encoding method, if the gain (pitch gain, algebraic codebook gain) is less than or equal to a certain threshold, the corresponding code (pitch lag code, algebraic code) Considering that the contribution to the voice is small, the index of the code is replaced with an arbitrary data sequence DT. For this reason,
In the voice code SCD1 of the one-encoding system, a section in which data is embedded and a section in which data is not embedded occur depending on the gain.

The embedding judging unit 121 judges whether or not another data is embedded in the code on a frame or sub-frame basis based on the gain of the speech code SCD1,
When it is determined that the data is embedded, the switch SW1 is closed and the voice code SCD1 is embedded in the data extraction unit 112.
To enter. The embedded data extraction unit 112 extracts data from the voice code SCD1 and stores the data in the FIFO buffer.
Enter in 2. The FIFO buffer is a first-in first-out buffer.

The embedding determination unit 123 determines whether to embed data in the voice code in frame or subframe units based on the gain of the voice code SCD2 'of the second coding system output from the code conversion unit 111, If it is determined that the data is to be embedded, the switch SW2 is closed, and the data holding unit 122 inputs the held data to the data embedding unit 113 in units of frames or subframes. As a result, the data embedding unit 113 embeds the data DT output from the data holding unit 122 in the voice code SCD2 ′ of the second encoding method in frame units and outputs it as the voice code SCD2.

The method of each embedding determination may be the same as the method used in each encoding method. When the embedding determination units 121 and 123 have different embedding determination methods, the closing timings of the switches SW1 and SW2 do not necessarily match. Further, even when the embedding determination method is the same, a similar phenomenon occurs because the voice code is different before and after conversion due to the conversion error of the voice code conversion unit 111. Figure
The third data holding unit 122 has a function of absorbing the above-mentioned difference in switching timing and preventing data loss.

That is, when the conversion destination is not the embedding target section, the data holding unit 122 temporarily holds the data DT extracted from the first voice code SCD1. Conversely, when the conversion source is not the embedding target section, the data held in the data holding unit 122 is extracted and embedded in the second voice code SCD2 '. Further, when the code data size of the embedding target of the conversion source is larger than the conversion destination, only the embeddable data amount is embedded, and the rest is temporarily held by the data holding unit 122. Further, when the number of held data in the data holding unit 122 decreases, the embedding of data at the conversion destination is temporarily stopped and the number of held data is restored. As described above, the difference in switching timing is absorbed to prevent data loss.

(B) Second system FIG. 4 is a conceptual diagram of a second system of the present invention. The communication system 101 of the conversion source has a voice line 105 and a data line 108, and the communication system 102 of the conversion destination is a voice system. The case where only the line 106 is provided is shown. As shown in the figure, the encoder 104 of the first encoding system in the communication system 101 encodes the input voice SP1 into a voice code SCD1 and sends the voice code to the voice line 105, and also outputs arbitrary data other than the voice code. The series DT is sent to the data line 108. Actually, the voice code SCD and the data sequence DT are time-division multiplexed, sent out to the multiplex line, separated at appropriate places, and input to the voice code conversion device 103. As described above, the voice code conversion device 103 receives the voice code from the voice line 105.
The data DT is input from the SCD1 and the data line 108, respectively. The voice code conversion device 103 is a voice code SC of the first coding method.
The D1 is converted into a voice code of the second encoding method, the data DT is embedded in the voice code, and the voice code SCD2 is transmitted to the communication system 102 of the conversion destination via the voice line 106.

The decoder 107 of the second coding system in the communication system 102 reads out and outputs an arbitrary data sequence DT embedded in the voice code, and at the same time, performs a normal decoder process on the voice code to reproduce the reproduced voice SP2. Output. At this time,
Since the embedding is performed so that the quality of the reproduced sound SP2 is hardly affected, the reproduced sound is almost the same as the case where the embedding is not performed. FIG. 5 is a configuration diagram of the code conversion apparatus 103 in the second system of the present invention, and the same parts as those of the code conversion apparatus in the first system of FIG. The difference is that the data DT is input via a route different from that of the voice code SCD1, and there is no embedded data extraction unit, and the embedded data DT is directly input to the data embedding unit 113.

The communication system as the conversion source time-division-multiplexes the voice code SCD1 and the data DT encoded according to the first encoding method and sends them to the multiplex line 200, and the line separating unit 201 outputs the voice code SCD1 and the data DT. Are separated and input to the code conversion device 103 via the voice line 105 and the data line 108. The data embedding unit 113, when the voice code SCD2 ′ of the second encoding method converted by the code conversion unit 111 and the data DT are input, embeds the data DT in the voice code SCD2 ′ in frame units,
The voice code SCD2 is sent to the voice line 106.

FIG. 6 is another block diagram of the code conversion device 103 in the second system of the present invention, in which the same parts as those of the code conversion device in the first system of FIG. 3 are designated by the same reference numerals. 3 is different from FIG. 3 in that the data DT is input via a route different from that of the voice code SCD1, there is no embedding determination unit, embedded data extraction unit, and the embedded data DT is directly input to the data holding unit 122. The communication system as the conversion source time-division-multiplexes the voice code SCD1 and the data DT encoded according to the first encoding method and sends them to the multiplex line 200, and the line separation unit 201 separates these voice code SCD1 and the data DT. Input to the code conversion device 103 via the voice line 105 and the data line 108.

The code conversion device 103 adaptively embeds the data DT in the voice code SCD 'based on the property of the voice code. That is, the code conversion unit 111 converts the speech code SCD1 of the first coding method into the speech code SCD2 'of the second coding method, and
The data holding unit 122 of the FO buffer structure receives the input data D
Hold T The embedding determination unit 123 determines whether to embed data in the voice code in frame or subframe units based on the voice code SCD2 ′ of the second encoding method output from the code conversion unit 111, and determines that the data is embedded. Then, the switch SW2 is closed, and the data holding unit 122 inputs the held data from the oldest one to the data embedding unit 113 in units of frames or subframes. As a result, the data embedding unit 113 embeds the data DT output from the data holding unit 122 in the voice code SCD2 ′ of the second coding method in frame units and sends it as the voice code SCD2 to the voice line 106.

(C) Third System FIG. 7 is a conceptual diagram of the third system of the present invention. Contrary to the second system, the communication system 101 of the conversion source is the voice line 1
05 only, the communication system 102 of the conversion destination is the voice line 106
And a data line 109 is shown. The encoder 104 of the first coding system in the communication system 101
SP1 is encoded, an arbitrary data series DT other than voice data is embedded in the code, and the voice code SCD1 is transmitted to the voice line 105. The voice code conversion device 103 converts the voice code SCD1 of the first coding method into the voice code SCD2 of the second coding method, extracts the data DT embedded in the voice code SCD1, and these voice code SCD2, Data DT for each line
Send to 106,109. The communication system 102 is a data line 109
The input data is output through the decoder 107, and the decoder 107 decodes the audio code SCD2 and outputs the reproduced audio SP2. Actually, the voice code SCD2 and the data DT are time division multiplexed at appropriate places, transmitted to the communication system 102, and separated by the communication system.

FIG. 8 is a block diagram of the code conversion device 103 in the third system of the present invention. The same parts as those of the code conversion device in the first system of FIG. 2 are designated by the same reference numerals.
The difference is that there is no data embedding unit, and the code conversion unit 111
The point that the data DT extracted by the embedded data extraction unit 112 is not embedded in the voice code SCD2 of the second encoding method that is output from
The point is that the data DT is transmitted separately from the voice code SCD2 of the second encoding method. Speech code SCD1 encoded according to the first encoding method at the conversion source and in which data DT is embedded
Are sequentially input to the code conversion unit 111 and the embedded data extraction unit 112 in frame units. The code conversion unit 111 converts the speech code SCD1 of the first coding method into the speech code SCD2 of the second coding method.
And is transmitted to the voice line 106. In addition, the embedded data extraction unit 112 uses the data D embedded in the voice code SCD1.
T is extracted and sent to the data line 109. Line multiplexer 203
Sends time-division-multiplexed voice code SCD2 and data DT input via voice line 106 and data line 109 to multiplex line 204.

FIG. 9 is another block diagram of the code conversion device 103 in the third system of the present invention, in which the same parts as those of the code conversion device in the first system of FIG. 3 is different from FIG. 3 in that there is no data holding unit, embedding determination unit, or data embedding unit, that data DT is not embedded in the voice code SCD2 output from the code conversion unit 111, and that the data DT is separate from the voice code SCD2. It is the point that is sent out.

The encoder of the communication system on the transmission side has a gain
When (pitch gain, algebraic codebook gain) is less than or equal to a certain threshold, the contribution of the corresponding code (pitch lag code, algebraic code) to speech is regarded as small, and the index of the code is replaced with an arbitrary data sequence DT. As a result, in the voice code SCD1 of the first encoding method, a section in which data is embedded and a section in which data is not embedded occur. The embedding determination unit 121 determines whether another data is embedded in the code on a frame or sub-frame basis based on the gain obtained from the voice code SCD1, and when it is determined that the data is embedded, The switch SW1 is closed and the voice code SCD1 is input to the embedded data extraction unit 112. The embedded data extraction unit 112 extracts embedded data from the voice code SCD1 and sends it to the data line 109. In parallel with the above, the voice code conversion unit 111 uses the voice code SCD1 of the first coding method.
Is converted into the voice code SCD2 of the second encoding system to convert the voice line 106
Send to. The line multiplexer 203 includes a voice line 106 and a data line 10.
The voice code SCD2 and the data DT input via 9 are time division multiplexed and transmitted to the multiplex line 204.

(B) Embodiment in First System (a) First Embodiment FIG. 10 is a block diagram of a code conversion device in the first system of the present invention, showing a configuration for embedding control. The first embodiment shows an example in which an AMR voice code in which arbitrary data is embedded is converted into a G.729A voice code without damaging the embedded data.
Furthermore, in the first embodiment, the encoder of the conversion source AMR embeds arbitrary data in all 17 bits / subframes allocated to the algebraic code if the algebraic codebook gain is smaller than the set value, and the algebraic code If the book gain is larger than the set value, the original algebraic code data is embedded. Similarly, the conversion destination G.729A encoder also embeds data in all 17 bits assigned to the algebraic code according to the algebraic codebook gain.

In FIG. 10, when the line data bst1 (m), which is the output of the AMR encoder of the m-th frame, is input to the code separation unit 114 through the terminal 1, the code separation unit 114 outputs the line data bst.
1 (m) is separated into element codes of AMR (LSP code 1, pitch lag code 1, pitch gain code 1, algebraic code 1, algebraic gain code 1). Then, these element codes are input to each code conversion unit (LSP code conversion unit 111a, pitch lag code conversion unit 111b, pitch gain code conversion unit 111c, algebraic gain code conversion unit 111d, algebraic code conversion unit 111e) in the code conversion unit 111. To do. Each of the code conversion units 111a to 111e converts the code of the first coding system into the code of the second coding system, but the operation thereof is the same as that of the conventional technique, and therefore the description thereof is omitted here. Below,
Only the part related to data embedding will be described.

The embedding judging unit 121 obtains an algebraic gain dequantized value (algebraic gain) from the algebraic gain code 1 and switches the switch SW1 according to the gain value. That is, when the AMR algebraic gain value is smaller than a certain threshold value, it is determined that there is embedded data, the switch SW1 is closed, and the algebraic code 1 is input to the embedded data extraction unit 112.
The embedded data extraction unit 112 extracts the embedded data Dcode included in the algebraic code and outputs it to the data holding unit 122. In the present embodiment, since data is embedded in all the AMR algebraic codes (17 bits / subframe), 17b
The data series of it is cut out as it is as embedded data Dcode. The data holding unit 122 having a FIFO structure stores and holds the input data series in the order of oldness.

On the other hand, the embedding determination unit 123 obtains an algebraic gain dequantized value from the converted G.729A algebraic gain code 2 input from the algebraic gain code conversion unit 111d, and switches SW2 according to the gain value. Switch. That is,
If the G.729A algebraic gain value is less than a certain threshold,
When it is determined that the data is to be embedded, the switch SW2 is closed and the data is input from the data holding unit 122 to the data embedding unit 113. In this embodiment, since data is embedded in all G.729A algebraic codes (17 bits / subframe), the data holding unit 122 inputs 17-bit data to the data embedding unit 113. The data embedding unit 113 embeds the input data in 17 bits assigned to the algebraic code 2. That is, all the G.729A algebraic codes (17 bits) are replaced with the data series (17 bits).

The algebraic code 2 in which the data is embedded is multiplexed by the code multiplexing unit 115 together with other element codes, and the line data of the nth frame of G.729A including the embedded data.
It is output from terminal 2 as bst2 (n). According to the first embodiment, when arbitrary data is embedded in the algebraic code in the AMR voice code bst1 (m), the data is embedded in the G.729A algebraic code without damaging the embedded data. It can be converted into a voice code bst2 (n). This enables data communication in addition to voice communication without changing the voice format between AMR and G.729A. Although the conversion from AMR to G.729A has been described above, the configuration of the portion related to the data extraction and data embedding in the first embodiment can be applied to the reverse conversion from G.729A to AMR.

(B) Second Embodiment FIG. 11 is another configuration diagram of the code conversion device in the first system of the present invention, showing the configuration for embedding control, which is the same as the first embodiment of FIG. The same parts are given the same reference numerals. The difference is that in the first embodiment, if the algebraic gain is smaller than the set value, arbitrary data is embedded in all 17 bits / subframes assigned to the algebraic code, but in the second embodiment, the pitch is changed. If the gain is smaller than the set value, it is assigned to the pitch lag code.
The point is that arbitrary data is embedded in all 8 bits or 5 bits / subframe.

The embedding determination unit 121 determines the pitch gain code 1
A pitch gain dequantized value (pitch gain) is obtained from the value, and the switch SW1 is switched according to the gain value. That is, when the pitch gain value of the AMR is smaller than a certain threshold value, it is determined that there is embedded data, the switch SW1 is closed, and the pitch lag code 1 is input to the embedded data extraction unit 112. The embedded data extracting unit 112 extracts the embedded data Dcode included in the pitch lag code and outputs it to the data holding unit 122. In this embodiment, the pitch lag code of AMR
Since the data is embedded in all (8 bits or 6 bits / subframe), the 8-bit or 6-bit data series is cut out as it is as the embedded data Dcode. The data holding unit 122 having a FIFO structure stores and holds the input data series in the order of oldness.

On the other hand, the embedding determination unit 123 obtains a pitch gain dequantized value from the converted pitch gain code 2 of G.729A input from the pitch gain code conversion unit 111c, and switches SW2 according to the gain value. Switch. That is, when the pitch gain value of G.729A is smaller than a certain threshold value, it is determined to embed the data, the switch SW2 is closed, and the data holding unit 122 stores the data in the data embedding unit 113.
To enter. In this embodiment, G.729A pitch lag code (8
Since data is embedded in all bits (5 bits or 5 bits / subframe), the data holding unit 122 stores the data of 8 bits or 5 bits according to the subframe.
To enter. The data embedding unit 113 uses the pitch lag code 2
The input data is embedded in the 8 bits or 5 bits allocated to.

Pitch lag code 2 with embedded data
Is multiplexed with other element codes by the code multiplexing unit 115 and output from the terminal 2 as the line data bst2 (n) of the n.th frame of G.729A including embedded data. According to the second embodiment, when arbitrary data is embedded in the pitch lag code of the AMR voice code bst1 (m), the voice in which the data is embedded in the G.729A pitch lag code without damaging the embedded data. It can be converted into the code bst2 (n). This enables data communication in addition to voice communication without changing the voice format between AMR (7.95 kbps) and G.729A. In the above, the conversion from AMR to G.729A has been described, but the configuration of the part related to data extraction and data embedding can be applied to the reverse conversion from G.729A to AMR and other code conversions. .

(C) Third Embodiment FIG. 12 is another block diagram of the code conversion device in the first system of the present invention, showing the structure in the case where the embedding control is not performed. The third embodiment shows an example in which the AMR voice code is converted into the G.729A voice code without damaging the embedded data. Figure 2 shows the voice code of AMR.
1 to FIG. 23, one frame is 20 msec, and 5 msec
4 sub-frames for each, 17 for each sub-frame
It has a bit algebraic sign. On the other hand, the voice code of G.729A is 10 msec per frame, two subframes are provided every 5 msec, and each subframe has a 17-bit algebraic code. For both AMR and G729A, the pulse position m0 to m3 and polarity s0 to s of four pulse systems (see Table 1) are set by these 17 bits.
3 is expressed. Bit assignments for the pulse positions m0 to m3 and the polarities s0 to s3 are as shown in FIG.

In the third embodiment, the conversion source AMR encoder is, for example, m3, s3 indicating the pulse position and polarity of the fourth path system.
Embed the data Dcode in 5 bits. The embedded data extracting unit 112 always stores the embedded data D included in the algebraic code 1.
The code is extracted and input to the data embedding unit 113. The data embedding unit 113 is assigned to the algebraic code 2 1
Data Dcod input to 5 bits of m3 and s3 out of 7 bits
Embed e. The algebraic code 2 in which the data is embedded is multiplexed with the other element codes in the code multiplexing unit 115, and the line data of the nth frame of G.729A including the embedded data.
It is output from terminal 2 as bst2 (n).

According to the first system described above, the first conversion source
The embedded data DT is once extracted from the audio code SCD1 of the encoding method, and the audio code SCD of the second encoding method after the code conversion is performed.
By embedding the data DT in 2 ′ again, the voice code S
The data DT embedded in the CD1 can be converted into the voice code SCD2 in which the data DT is embedded without damaging the same.
Further, according to the first system, when the embedding control is adaptively performed between the conversion source and the conversion destination, due to the difference in the embedding control method of each coding method or the conversion error in the conventional speech code conversion unit. By absorbing the generated timing difference between the data extraction and the embedding by the data holding unit, it is possible to convert the data embedded in the voice code SCD1 into the voice code SCD2 in which the data is embedded. Further, according to the first system, between voice communication systems having a voice line to which the data embedding technology is applied, the voice data is transmitted via the voice line without damaging the embedded data and without changing the voice code format. It becomes possible to perform both voice and data communication.

(C) Embodiment of Second System of the Present Invention (a) First Embodiment FIG. 14 is a block diagram of a voice code conversion device in the second system of the present invention, wherein voice code bst1 (m) Unlike the first system embodiment, the data Dcode is not embedded in and the data is input to the voice code conversion device via a separate line from the voice code. The line multiplexer 201 separates the voice code bst1 (m) and the data Dcode from the multiplex data received via the multiplex line 200, inputs the voice code bst1 (m) from the terminal 1 to the code separator 114, and from the terminal 3. The data Dcode is directly input to the data holding unit 122. The code separation unit 114 converts the line data bst1 (m) into the element code of AMR.
(LSP code 1, pitch lag code 1, pitch gain code 1, algebraic code 1, algebraic gain code 1) and these element codes are code conversion units in the code conversion unit 111 (LSP code conversion unit 1
11a, pitch lag code conversion unit 111b, pitch gain code conversion unit 111c, algebraic gain code conversion unit 111d, algebraic code conversion unit
Enter it into 111e). Each of the code conversion units 111a to 111e converts the code of the first coding method into the code of the second coding method.

The embedding determination unit 123 obtains an algebraic gain dequantization value from the converted G.729A algebraic gain code 2 input from the algebraic gain code conversion unit 111d, and switches the switch SW2 according to the gain value. I do. That is, G.729
When the algebraic gain value of A is smaller than a certain threshold value, it is determined to embed the data, the switch SW2 is closed, and the data is input from the data holding unit 122 to the data embedding unit 113.
The data embedding unit 113 embeds the input data in 17 bits assigned to the algebraic code 2. The algebraic code 2 in which the data is embedded is multiplexed with the other element codes in the code multiplexing unit 115, and G.7 containing the embedded data is included.
Terminal 2 as line data bst2 (n) of the 29th nth frame of 29A
Will be output.

According to this embodiment, in the case where the communication system on the AMR side has a data line in addition to a voice line,
G.729A communication system that has only voice line by converting voice code bst1 (m) and data Dcode input separately via voice line and data line to voice code bst2 (n) with embedded data Can be transmitted to. As a result, from a communication system capable of voice communication and data communication, for example, a third-generation mobile phone system (AMR is adopted as a voice encoding method), a communication system having only a voice line, for example, a conventional second-generation system that performs only voice communication. It becomes possible to perform data communication in addition to voice communication to the mobile phone system (G.729A).

(A) Second Embodiment FIG. 15 is another block diagram of the speech code conversion apparatus in the second system of the present invention, showing the configuration when the embedding control is not performed. In the second embodiment, the voice code bs
The data Dcode is not embedded in t1 (m), and the data is input to the voice code conversion device via a line different from the voice code.
Further, the G729A algebraic code expresses each pulse position m0 to m3 and polarity s0 to s3 of four pulse systems by 17 bits,
In the second embodiment, for example, the data Dcode is embedded in 5 bits of m3 and s3 indicating the pulse position and polarity of the fourth pass system. The line multiplexer 201 separates the voice code bst1 (m) and the data Dcode from the multiplex data received via the multiplex line 200, inputs the voice code bst1 (m) from the terminal 1 to the code separator 114, and from the terminal 3. The data Dcode is directly input to the data embedding unit 113. The code separation unit 114 converts the line data bst1 (m) into the element code of AMR.
(LSP code 1, pitch lag code 1, pitch gain code 1, algebraic code 1, algebraic gain code 1) and these element codes are code conversion units in the code conversion unit 111 (LSP code conversion unit 1
11a, pitch lag code conversion unit 111b, pitch gain code conversion unit 111c, algebraic gain code conversion unit 111d, algebraic code conversion unit
Enter it into 111e). Each of the code conversion units 111a to 111e converts the code of the first coding method into the code of the second coding method.

The data embedding unit 113 embeds the input data Dcode in 5 bits of m3 and s3 among the 17 bits assigned to the algebraic code 2. The data-embedded algebraic code 2 is multiplexed with other element codes by the code multiplexing unit 115, and is output from the terminal 2 as the G.729A nth frame line data bst2 (n) including the embedded data. It

According to the second system described above, it is possible to perform voice communication and data communication without changing the voice code format from a communication system having a data line separate from the voice line to a communication system having only a voice line. Becomes
In the above, the conversion from AMR to G.729A has been described.
It can be applied at the time of reverse conversion from 729A to AMR and at the time of other code conversions. Further, although the case where data is embedded in the algebraic code according to the algebraic gain has been described above, data may be embedded in the pitch lag code according to the pitch gain.

(D) Third system of the present invention (a) First embodiment FIG. 16 is a block diagram of a speech code conversion apparatus in the third system of the present invention, in the case of adaptively extracting embedded data. Shows the configuration of. In this embodiment, the first coding method is G.729A and the second coding method is AMR (7.95 kbps).
The code conversion device converts the G.729A voice code into an AMR voice code and transmits the voice code, and also extracts the data embedded in the G.729A voice code and transmits the data separately from the voice code. Further, the conversion source G.729A encoder (not shown) is
If the algebraic gain is smaller than the set value, arbitrary data is embedded in all 17 bits / subframes allocated to the algebraic code, and if the algebraic gain is larger than the set value, the original algebraic code data is embedded.

When the line data bst1 (m), which is the output of the G.729A encoder of the mth frame, is input to the code separating unit 114 through the terminal 1, the code separating unit 114 sets the line data bst1 (m) to G.729A. 7
The 29A element code (LSP code 1, pitch lag code 1, pitch gain code 1, algebraic code 1, algebraic gain code 1) is separated.
Then, these element codes are converted into code conversion units in the code conversion unit 111 (LSP code conversion unit 111a, pitch lag code conversion unit 111.
b, pitch gain code conversion unit 111c, algebraic gain code conversion unit 111d, and algebraic code conversion unit 111e). Each of the code conversion units 111a to 111e converts the G.729A code into an AMR code, and the code multiplexing unit 115 multiplexes each AMR code to produce a voice code bst2 (n).
To the line multiplexer 203.

In parallel with the above, the embedding judging unit 121 determines the algebraic gain code 1 to the algebraic gain dequantized value (algebraic gain).
Then, the switch SW1 is switched according to the gain value. That is, when the algebraic gain value of G.729A is smaller than a certain threshold value, it is determined that there is embedded data, the switch SW1 is closed, and the algebraic code 1 is set as the embedded data extraction unit 112.
To enter. The embedded data extraction unit 112 extracts the embedded data Dcode included in the algebraic code, and the line multiplexing unit 203
To enter. Since the data is embedded in all G.729A algebraic codes (17 bits / subframe), 17 bits
The data series of is cut out as it is as embedded data Dcode and is input to the line multiplexing unit 203. The line multiplexer 203 multiplexes the input voice code bst2 (n) and the data Dcode and sends them to the multiplex line 204.

(B) Second Embodiment FIG. 17 is another block diagram of the speech code conversion apparatus in the third system of the present invention, in which the embedded data is always inserted in the algebraic code. In this example, the first
The encoding method is G.729A, and the second encoding method is AMR (7.95kbp
s), and the voice transcoding device uses the G.729A voice code for AMR.
The G.729A voice code is extracted and transmitted on a separate line from the voice code. In addition, it is assumed that the G.729A encoder that is the conversion source embeds the data Dcode in 5 bits (see FIG. 13) of m3 and s3 of the algebraic code.

When the line data bst1 (m), which is the G.729A encoder output of the mth frame, is input to the code separation unit 114 through the terminal 1, the code separation unit 114 sets the line data bst1 (m) to G.729A. 7
The 29A element code (LSP code 1, pitch lag code 1, pitch gain code 1, algebraic code 1, algebraic gain code 1) is separated.
Then, these element codes are converted into code conversion units in the code conversion unit 111 (LSP code conversion unit 111a, pitch lag code conversion unit 111.
b, pitch gain code conversion unit 111c, algebraic gain code conversion unit 111d, and algebraic code conversion unit 111e). Each of the code conversion units 111a to 111e converts the G.729A code into an AMR code, and the code multiplexing unit 115 multiplexes each AMR code to produce a voice code bst2 (n).
To the line multiplexer 203.

In parallel with the above, the embedded data extraction unit 112
Extracts the embedded data Dcode included in the algebraic code and inputs it to the line multiplexer 203. Since the data is embedded in the G.729A algebraic code m3, s3 bit positions, the data is cut and input to the line multiplexer 203 as embedded data Dcode. The line multiplexer 203 multiplexes the input voice code bst2 (n) and the data Dcode and sends them to the multiplex line 204. According to the third system, it becomes possible to perform voice communication and data communication without changing the voice code format from the communication system having only the voice line to the communication system having the data line separately from the voice line. Above, G.729A
→ The conversion to AMR was explained, but it can be applied to other code conversions. Further, although the case where data is embedded in the algebraic code according to the algebraic gain has been described above, data may be embedded in the pitch lag code according to the pitch gain.

Supplementary note (Supplementary note 1) In the speech code conversion method for converting the first speech code obtained by coding the input speech by the first speech coding method into the second speech code by the second speech coding method, the first speech When arbitrary data is embedded in the code, the first voice code is converted into the second voice code, embedded data is extracted from the first voice code, and the first voice code obtained by the conversion is extracted.
(2) A voice code conversion method comprising embedding the extracted data in a voice code. (Supplementary Note 2) When data is embedded in the first voice code by replacing a part of the first voice code with the data when the data embedding condition is satisfied at the transmission source,
The dequantized value of a predetermined element code forming the received first voice code is referred to monitor whether the data embedding condition is satisfied, and if the data embedding condition is satisfied, the first voice code is used to embed the data. The voice code conversion method according to appendix 1, wherein data is extracted. (Supplementary note 3) The voice code conversion method according to supplementary note 2, wherein the extracted embedded data is stored in a data holding unit, and the embedded data is read from the data holding unit and embedded in the second voice code. (Supplementary Note 4) When data is embedded in the first voice code by replacing part of the first voice code with the data when the data embedding condition is satisfied at the transmission source,
By referring to the dequantized value of a predetermined element code constituting the first voice code received from the transmission source, it is monitored whether the data embedding condition is satisfied, and if the data embedding condition is satisfied, the first voice The embedded data is extracted from the code, the extracted embedded data is held, and the data embedding condition is satisfied by referring to the dequantized value of the predetermined element code that constitutes the second speech code obtained by the conversion. The voice code as set forth in appendix 1, characterized in that if it is satisfied, the data is embedded in the second voice code by replacing a part of the second voice code with the held data. How to convert. (Supplementary note 5) In the voice code conversion method of converting the first voice code obtained by encoding the input voice by the first voice encoding method into the second voice code by the second voice encoding method, transmitting the first voice code and data. Separately received from the original, the first voice code to the second
A voice code conversion method comprising converting to a voice code, embedding the data in a second voice code obtained by the conversion, and transmitting the data to a destination. (Supplementary Note 6) The first voice code is received from a voice line, the data is received from a data line, and the second voice code in which the data is embedded is transmitted to a destination via the voice line. A method of converting a voice code according to appendix 5. (Supplementary Note 7) The received data is stored in a data holding unit, and by referring to the dequantized value of a predetermined element code forming the second speech code, it is monitored whether the data embedding condition is satisfied, and the condition is satisfied. The voice code conversion method according to appendix 5, wherein the data is embedded in the second voice code by replacing a part of the second voice code with the data stored in the data holding unit. (Supplementary Note 8) In a voice code conversion method for converting a first voice code obtained by encoding an input voice by a first voice encoding method into a second voice code by a second voice encoding method, receiving a first voice code, When arbitrary data is embedded in the first voice code, the first voice code is converted into a second voice code, and the embedded data is extracted from the first voice code to obtain a second voice code. A voice code conversion method comprising transmitting a voice code and the extracted data separately to a destination. (Supplementary Note 9) When data is embedded in the first voice code by replacing part of the first voice code with the data when the data embedding condition is satisfied at the transmission source,
By referring to the dequantized value of a predetermined element code constituting the received first voice code, it is monitored whether the data embedding condition is satisfied, and if the data embedding condition is satisfied, the first voice code is used. 9. The audio code conversion method according to appendix 8, wherein embedded data is extracted. (Supplementary note 10) In a voice code conversion device for converting a first voice code obtained by encoding an input voice according to a first voice encoding system into a second voice code according to a second voice encoding system, arbitrary data is used as the first voice code. Is embedded, a code conversion unit that converts the first voice code into a second voice code, an embedded data extraction unit that extracts embedded data from the first voice code, and a second voice code obtained by the conversion. A voice code conversion device comprising a data embedding unit for embedding the extracted data. (Supplementary Note 11) When data is embedded in the first voice code by replacing a part of the first voice code with the data when the data embedding condition is satisfied at the transmission source, the embedded data extraction unit receives the data. It is monitored whether the data embedding condition is satisfied by referring to the dequantized value of a predetermined element code that constitutes the first voice code, and if the data embedding condition is satisfied, the embedded data is sent from the first voice code. The voice code conversion device according to appendix 10, wherein the voice code conversion device according to claim 10 is extracted. (Supplementary Note 12) A data storage unit for storing the extracted embedded data is further provided, wherein the embedded data extraction unit stores the extracted embedded data in the data storage unit and the data embedding unit stores the extracted data in the data storage unit. 12. The voice code conversion device according to appendix 11, wherein the embedded data is read out and embedded in the second voice code. (Supplementary Note 13) The embedded data extraction unit is configured to operate in the second
By referring to the dequantized value of a predetermined element code forming the voice code, it is monitored whether the data embedding condition is satisfied, and if the condition is satisfied, the second voice code is stored in the data holding unit. 13. The voice code conversion apparatus according to appendix 12, wherein the data is embedded in the second voice code by replacing a part of the above. (Supplementary Note 14) In a voice code conversion device for converting a first voice code obtained by encoding an input voice by a first voice encoding method into a second voice code by a second voice encoding method, transmitting a first voice code and data Receiving means for separately receiving from the original, a code conversion unit for converting the first voice code into a second voice code, a data embedding unit for embedding the data in the second voice code obtained by the conversion and transmitting it to the destination, A voice code conversion device comprising: (Supplementary Note 15) The voice code conversion device further includes a data holding unit for storing the data, and the data embedding unit refers to an inverse quantized value of a predetermined element code forming the second voice code and sets a data embedding condition. And a means for embedding the data in the second voice code by replacing a part of the second voice code with the data stored in the data holding unit. 14. The voice code conversion device according to appendix 14, characterized in that. (Supplementary Note 16) In a voice code conversion device for converting a first voice code obtained by encoding an input voice by a first voice encoding method into a second voice code by a second voice encoding method, a first voice received from a transmission source. When arbitrary data is embedded in the code, a code conversion unit that converts the first voice code into a second voice code, an embedded data extraction unit that extracts embedded data from the first voice code, and is obtained by the conversion. A voice code conversion device comprising: a means for separately transmitting a second voice code and the extracted data to a destination. (Supplementary Note 17) When the data embedding condition is satisfied at the transmission source, when the data is embedded in the first voice code by replacing a part of the first voice code with the data, the embedded data extraction unit, 1 received from source
By referring to the dequantized value of a predetermined element code constituting the voice code, it is monitored whether the data embedding condition is satisfied, and if the data embedding condition is satisfied, the embedded data is extracted from the first voice code. The speech code conversion device according to appendix 16, characterized in that.

[0078]

As described above, according to the present invention, the embedded data is once extracted from the voice code of the first encoding method of the conversion source, and the data is re-converted to the voice code of the second encoding method after the code conversion. By embedding, it is possible to convert the data embedded in the voice code of the first encoding method into the voice code of the second encoding method in which the same data is embedded, without damaging the data. Further, according to the present invention, when the embedding control is adaptively performed between the conversion source and the conversion destination, data generated due to a difference in the embedding control method of each encoding method or due to a conversion error in the conventional speech code conversion unit. By absorbing the difference between the extraction timing and the embedding timing by the data holding unit, the data embedded in the voice code of the first encoding system is converted into the voice code of the second encoding system in which the same data is embedded. can do.

Further, according to the present invention, between voice communication systems having a voice line to which the data embedding technique is applied, the voice data is transmitted via the voice line without damaging the embedded data and without changing the voice code format. It becomes possible to communicate both voice and data. Further, according to the present invention, when the voice code and the data of the first encoding method are input to the voice code conversion unit via a separate line from the system of the conversion source, the voice code conversion unit performs the second code after the code conversion. By embedding the data in the voice code of the conversion system, it becomes possible to transmit to the conversion destination only by the voice line. Further, according to the present invention, when the voice code of the first coding method in which the arbitrary data DT is embedded is input from the system of the conversion source via the voice line, the voice code conversion unit converts the voice code from the voice code. The embedded data is extracted and sent to the data line, and the voice code of the first coding method is converted into the voice code of the second coding method and sent to the voice line, so that it is transmitted by the voice line of the conversion source. The voice information and the data information can be separately transmitted to the voice line and the data line of the conversion destination.

Further, according to the present invention, voice communication and data communication can be performed between a communication system having only a voice line and a communication system having a data line separately from the voice line without changing the voice code format. It will be possible. In the future, due to the spread of multimedia information communication, data will be used in communication between various communication systems such as communication between conventional mobile phone systems and next-generation mobile phone systems, or communication between VoIP and mobile systems such as mobile phones. The effect of the present invention is great because there is a high need for a technology that uses both the embedding technology and the voice code conversion technology.

[Brief description of drawings]

FIG. 1 is a first system conceptual diagram of the present invention.

FIG. 2 is a configuration diagram of a voice code conversion device in the first system of the present invention.

FIG. 3 is another schematic configuration diagram of the speech code conversion apparatus in the first system of the present invention.

FIG. 4 is a second system conceptual diagram of the present invention.

FIG. 5 is a schematic configuration diagram of a speech code conversion device in the second system of the present invention.

FIG. 6 is another schematic configuration diagram of the speech code conversion device in the second system of the present invention.

FIG. 7 is a conceptual diagram of a third system according to the present invention.

FIG. 8 is a schematic configuration diagram of a voice code conversion device in a third system of the present invention.

FIG. 9 is another schematic configuration diagram of the speech code conversion apparatus in the third system of the present invention.

FIG. 10 is a configuration diagram of a voice code conversion device in the first system of the present invention.

FIG. 11 is a configuration diagram of another embodiment of the voice code conversion device in the first system of the present invention.

FIG. 12 is a configuration diagram of still another embodiment of the voice code conversion device in the first system of the present invention.

FIG. 13 is a configuration diagram of an algebraic code.

FIG. 14 is a configuration diagram of an embodiment of a voice code conversion device in the second system of the present invention.

[Fig. 15] Fig. 15 is a configuration diagram of another embodiment of the voice code conversion device in the second system of the present invention.

FIG. 16 is a configuration diagram of an embodiment of a voice code conversion device in the third system of the present invention.

[Fig. 17] Fig. 17 is a configuration diagram of another embodiment of the speech code conversion device in the third system of the present invention.

[Fig. 18] Fig. 18 is a configuration diagram of an ITU-T recommended G.729A system encoder.

FIG. 19 is an explanatory diagram of sample points assigned to each pulse system group 1 to 4.

FIG. 20 is a block diagram of a G.729A system decoder.

[Fig. 21] Fig. 21 is an explanatory diagram of comparison between main specifications of the G.729A system and AMR.

[Fig. 22] Fig. 22 is an explanatory diagram of frame configurations of the G.729A system and AMR.

[Fig. 23] Fig. 23 is a diagram for explaining comparison of bit allocation in the G.729A system and the AMR system.

FIG. 24 is an explanatory diagram of voice code conversion between different communication systems.

[Fig. 25] Fig. 25 is an explanatory diagram of a conventional technique for converting a voice code into a code of another voice encoding method.

FIG. 26 is a conceptual diagram of a voice communication system to which a data embedding technique is applied.

FIG. 27 is a principle diagram of code conversion.

[Explanation of symbols]

103 code converter 111 code conversion unit 112 Embedded data extractor 113 Data embedding section

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁷ Identification code FI theme code (reference) H04B 14/04 G10L 9/18 A (72) Inventor Kyoji Ohta 4 chome Uedanaka, Nakahara-ku, Kawasaki-shi, Kanagawa 1-1 In Fujitsu Limited (72) Inventor Masanao Suzuki 4-1-1 Kamiotanaka, Nakahara-ku, Kawasaki-shi, Kanagawa Prefecture 1-1 1-1 Masayoshi Tanaka Inventor Masataka Tanaka 4-chome, Ueda-anaka, Kawasaki-shi, Kanagawa No. 1 No. 1 F term in Fujitsu Limited (reference) 5D045 CA02 CC03 CC07 DA11 5J064 AA01 BB03 BC01 BC26 BD02 5K041 BB08 CC01 HH21 HH22

Claims (5)

[Claims] 1. A voice code conversion method for converting a first voice code obtained by encoding an input voice by a first voice encoding method into a second voice code according to a second voice encoding method, wherein the first voice code is arbitrary. If the data is embedded,
A voice characterized by converting the first voice code to a second voice code, extracting embedded data from the first voice code, and embedding the extracted data in a second voice code obtained by the conversion. Code conversion method. 2. When data is embedded in the first voice code by replacing a part of the first voice code with the data when the data embedding condition is satisfied at the source, the first voice code received from the source (1) By referring to the dequantized value of a predetermined element code that constitutes one voice code, monitor whether the data embedding condition is satisfied, and if the data embedding condition is satisfied, the embedded data from the first voice code Extraction is carried out, the extracted embedded data is held, and it is monitored whether the data embedding condition is satisfied by referring to the dequantized value of the predetermined element code that constitutes the second speech code obtained by the conversion, and it is satisfied. The voice code according to claim 1, wherein, in the case where the voice code is stored, the data is embedded in the second voice code by replacing a part of the second voice code with the held data. No. conversion method. 3. A voice code conversion method for converting a first voice code obtained by encoding input voice by a first voice encoding method into a second voice code according to a second voice encoding method, wherein the first voice code and data are A voice code characterized by receiving separately from a transmission source, converting a first voice code into a second voice code, embedding the data in a second voice code obtained by the conversion, and transmitting the data to a destination. How to convert. 4. A voice code conversion method for converting a first voice code obtained by encoding an input voice by a first voice encoding method into a second voice code according to a second voice encoding method, wherein the first voice code is received. , If any data is embedded in the first voice code, the first voice code is converted into a second voice code, embedded data is extracted from the first voice code, and the first voice code is obtained by the conversion. (2) A voice code conversion method, comprising transmitting the voice code and the extracted data separately to a destination. 5. A voice code conversion device for converting a first voice code obtained by encoding an input voice by a first voice encoding system into a second voice code according to a second voice encoding system, wherein the first voice code is converted into an arbitrary first voice code. If the data is embedded,
A code conversion unit that converts the first voice code into a second voice code, an embedded data extraction unit that extracts embedded data from the first voice code, and a data embedding that embeds the extracted data in the second voice code obtained by the conversion. A voice code conversion device comprising: