JPH03245199A - Error compensating system - Google Patents

Abstract

PURPOSE:To obtain a natural voice with less noises even in environment where the error rate of a transmission line is inferior by reproducing a voice signal by using the parameter of the synthesizing filter of a frame where an error correction is made and a driving signal. CONSTITUTION:A driving signal segmenting means 180 segments the signal for one pitch period from the accumulated driving signal when there is a specified or more error. A driving signal forming means 190 forms the driving signal for one frame by using the segmented signal for one pitch period. The synthesizing filter 200 is inputted with the formed driving signal and reproduces the voice signal of the frame where there is the specified or more error. An output means 220 outputs the voice signal reproduced when the specified or more error is detected. This means outputs a decoded voice signal when the specified or more error is not detected. The natural voice signal of the less noises is reproduced in this way even if there are many transmission line errors.

Description

DETAILED DESCRIPTION OF THE INVENTION [Object of the Invention] (Industrial Application Field) The present invention relates to an error compensation method used to suppress deterioration of sound quality due to line errors in a mobile communication system such as a car telephone.

(Prior Art) In mobile communication systems such as car telephones, there is a problem in which bit errors occur on the propagation path, resulting in significant deterioration of voice quality. Therefore, in order to suppress the deterioration of audio quality due to bit errors, there is a method called "decoding device J (+(]-2
0'1827). This content will be explained below.

FIG. 7 is a block diagram showing the configuration of a decoding device that includes the error compensation method described herein.

In FIG. 7, 501 is an input terminal, which is connected to an error correction section 502, a claim period detection section 504, and an error rate measurement section 506. The error correction section 502 is connected to a decoder 503, and the decoder 503 is connected to a variable amplitude limiter 50.
7, and the variable width limiter 507 is connected to the output end 510. The frame period detection section 504 is connected to an error correction section 502, a decoder 503, and an error rate measurement section 506, and the error rate measurement section 506 is connected to a variable amplitude limiter 507.

Next, the operation of the above embodiment will be explained. In the above embodiment, from the audio data input from the input terminal 501,
A frame period detection unit 504 obtains frame timing. The error correction unit 502 corrects errors in audio data input from the input terminal 501 for each frame, and the decoder 503 decodes the error-corrected audio data and outputs an audio signal. The error rate measuring section 506 measures the error rate of the transmission path from the frame period bits of the audio data inputted from the input terminal 501, and controls the variable amplitude limiter 507. When the error rate of the transmission path is lower than a certain value, the variable amplitude limiter 507 outputs the decoded audio output from the decoder 503 as it is to the output terminal 510, and when the error rate is higher than the certain value, the decoder 503 outputs it. The decoded audio is amplitude limited and output to the output terminal 510.

As mentioned above, even with the conventional error compensation method, when the error rate of the transmission path becomes high, the amplitude of the output voice is limited, so that the noise level due to transmission path errors can be reduced.

(Problem to be solved by the invention) However, in the conventional method described above, when the error rate becomes high, the decoding output is simply reduced, so not only the noise but also the voice level becomes very low and it becomes mumbled. There was a situation where the sound became discontinuous, making it unpleasant and unpleasant.

The present invention was made to solve these conventional problems, and it is an object of the present invention to provide an error compensation method that can reproduce natural audio signals with less noise even when there are many transmission path errors. shall be.

[Structure of the Invention (Means for Solving the Problems)] In order to achieve the object listed in L, the present invention provides a system in which a coded signal sequence coded for each fixed-length or variable-length frame is manually generated for each frame. means, error correction means for correcting errors in the coded signal sequence, error detection means for detecting errors in the coded signal sequence, signal means for decoding the error-corrected coded signal sequence; means for accumulating parameters of a synthesis filter stored in a decoding means, means for accumulating a drive signal for the synthesis filter obtained by the decoding means, and means for obtaining a pitch period based on the decoded signal obtained from the decoding means; means for cutting out a drive signal corresponding to a pitch period from the drive signal stored in the drive signal storage means when the error detection means detects an error of a certain value or more; and one frame using the cut out drive signal. If the error detection means does not detect an error of a certain value or more,
The present invention provides an error compensation system having means for outputting the audio signal decoded by the decoding means.

(Operation) The input means establishes frame synchronization, inputs the encoded signal sequence for each frame, and outputs the encoded signal sequence to the error correction means and the error detection means. The error correction means performs error correction on the encoded signal sequence, and the error detection means detects whether or not the encoded signal sequence contains more than a certain number of errors. The decoding means decodes the decoded signal sequence and outputs synthesis filter parameters, a synthesis filter drive signal, and a decoded audio signal. The means for accumulating the parameters of the synthesis filter and the means for accumulating the driving signal respectively accumulate the parameters and the driving signal of the frames in which the encoded signal sequence does not contain more than a certain error, and update the parameters and the driving signal for each frame in which the encoded signal sequence does not contain more than a certain error. .

The pitch period obtaining means obtains the bi-sochi period of the decoded signal and outputs it to the drive signal extraction means 2.

The drive signal cutting means cuts out a signal for one pitch period from the accumulated drive signal when there is an error of -window or more, and the drive signal generating means cuts out a signal for one pitch period from the accumulated drive signal, and uses the cut out signal for one pitch period. Generate drive signals for frames. The synthesis filter inputs the generated drive signal and reproduces the audio signal of the frame in which the error is greater than -window. The output means outputs a reproduced audio signal when an error larger than a window is detected, and outputs a decoded audio signal when an error larger than a -window is not detected.

(Embodiments) Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.

FIG. 1 and FIG. 2 are block diagrams of an encoding device and a decoding device, respectively, according to an embodiment of the present invention.

In FIG. 1, reference numeral 10 denotes an input terminal to which a digitized audio signal sequence is manually input. The encoded audio signal sequence is first encoded in the encoding section 20, then an error correction code is added in the error correction encoding section 30, and then output to the buffer 40. Can various known codes be used as the error correction code? Since burst errors occur in mobile communication, a Reed-Solomon code with a high ability to correct burst errors is used in this embodiment. A buffer 40 stores two frames worth of encoded signal sequences and connects them to the interleaving section. For example, as shown in FIG. 3, the increase unit constructs the encoded signal sequence of frame B' from a part of the encoded signal sequence of frame A and a part of the encoded signal sequence of frame B. A frame for transmission is constructed by exchanging the encoded signal sequences of the two frames. By doing this, burst errors in one frame are randomized into two frames, so deterioration in the quality of synthesized speech due to line errors can be reduced. 60 is an output terminal of the encoding device. The configuration and operation of the encoding unit 20 will be described later. Next, the decoding device will be explained.

In FIG. 2, a buffer 110 stores two frames of the decoded signal sequence input from the input terminal 100. The deinterleaving unit 120 performs the opposite process to that of the increment unit 50, and in FIG.
The encoded signal sequence of frame A is constructed from the following. The error correction unit 130 performs Reed-Solomon error correction decoding on the encoded signal sequence for each frame to correct errors in the decoded signal sequence. At the same time, the syndrome is calculated and output to the error detection section 140. Error correction codes are explained in detail in ``Coding Theory'' (written by Hiroka, published by Corona Publishing).

The error detection section 140 calculates the number of erroneous symbols from the syndrome calculated by the error correction section 130, and when the number is greater than the error correction ability, it is set as Error-1, and when it is below, it is set as Err.
ror-0 is output to the parameter memory 160, drive signal memory 170, extraction circuit 180, and switch circuit 210. The decoding unit 150 decodes the error-corrected encoded signal and stores the parameters of the synthesis filter in the parameter memory 1.
60 and a pitch period which is part of the parameters of the synthesis filter are output to the extraction circuit 180, the drive signal of the synthesis filter is output to the drive signal memory 170, and the decoded audio signal is output to the switch circuit 210.

The configuration and operation of the decoding section will be described later.

When the output of the error detection section 140 is Error-0, the parameter memory 160 stores the parameters of the synthesis filter, which is the output of the decoding section 150, without updating them. Similarly, when Error=00, the drive signal memory 170 stores the drive signal output from the decoder without updating it. The extraction circuit 180 detects whether the output of the imposition detection unit 140 is
When = 1, a signal corresponding to one pitch cycle sample is extracted from the last sample store (the frame immediately before Error = 1) of the drive signal stored in the drive signal memory 170, and is sent to the drive signal generation circuit]90. Output. The drive signal generation circuit 190 generates a drive signal for one frame by repeatedly connecting drive signals for one pitch period, and outputs the drive signal to the synthesis filter 200. Synthesis filter 2
00 manually generates a drive signal generated by the drive signal generation circuit 190, synthesizes an audio signal using the parameters stored in the parameter memory 160, that is, the parameters of the error-corrected frame, and synthesizes the audio signal to the switch circuit 210.
Output to. The switch circuit 210 switches the decoding unit 15 when the error is within the correctable range (Error=0).
The decoded audio signal, which is the output of
), the synthesized speech signal that is the output of the synthesis filter 200 is output to 220. Next, the configuration and operation of the encoding section and decoding section in FIGS. 1 and 2 will be explained.

FIG. 4 and FIG. 5 are block diagrams showing exemplary configurations of an encoding section and a decoding section, respectively.

In FIG. 4, the frame buffer 311 is connected to input terminal 3.
10, each block in FIG. 4 is a frame buffer 311.
The following processing is performed for each frame or subframe using .

The predicted parameter calculation circuit 312 calculates the predicted igI parameters using a known method. When the prediction filter 314 is configured by continuously connecting a long-term prediction filter 351 and a short-time prediction filter 352 as shown in FIG. 6, the prediction parameter calculation circuit 312 calculates the pitch period, pitch prediction coefficient, linear prediction coefficient, (α parameter or α parameter) is calculated using a known method such as an autocorrelation method or a covariance method. The calculation method is described, for example, in the above-mentioned document 2 (Sadateru Furui, Digital Speech Processing J, published by Tokai University Press in 19g5).The calculated prediction parameters are input to the prediction parameter encoding circuit 313. .

The prediction parameter encoding circuit 313 encodes the prediction parameter based on a predetermined number of quantization bits, outputs this code to the multiplexer 325, and sends the decoded value to the prediction filter 314, the synthesis filter 315, and the perceptual weighting filter 320. Output to. Prediction filter 314
calculates a prediction residual signal using the input audio signal and prediction parameters as input, and sends it to the density pattern selection circuit 315.
Output to.

In this embodiment, the density pattern selection circuit 315 first divides the prediction residual signal of one frame into a plurality of subframes, and calculates the sum of squares of the prediction residual signals of each subframe. Next, the density (pulse interval) pattern of the drive pulse train signal in the subframe is determined based on the sum of squares of the prediction residual signal. One example of a specific method is to set in advance two types of density patterns, pulse interval or shortest, and the number of subframes with a long pulse interval, and then pulse the pulses in the order of the subframes with the largest sum of squares of the prediction residual signal. This is a method of selecting a density pattern with shorter intervals.

The gain calculation circuit 327 inputs information on the selected density pattern and calculates the gain of the drive signal by, for example, 2 using the standard deviation of the prediction residual signals of all subframes with short pulse intervals.
Find the type. The obtained density pattern and gain are respectively encoded by the encoding circuit 316328 and inputted to the multiplexer 325, and their decoded values are inputted to the drive signal generation circuit 3]7. Drive signal generation circuit 317
are the density pattern and gain input from the encoding circuits 316 and 328, the normalized amplitude of the drive pulse input from the codebook 324, and the phase search circuit 322.
Based on the phase of the drive pulse input from the subframe, a drive signal with variable density is generated in units of subframes.

The key (a+) of the drive pulse in the m-th subframe is G, and the normalized amplitude of the drive pulse is ゛(1
) The number of pulses is Q, the phase of the pulse is K, g l
I, l
Letting the length of the subframe be L, the drive signal ex'''(n) can be described by the following equation.

t (n - (i-1) D -K l
(1) ffl G n-1, 2, . . . , L] ≦ K 5D lIl Note that the phase K is the leading position of the pulse in the subframe. Further, (n) is the Kronesoka delta function.

The drive signal generated by the drive signal generation circuit 317 is input to a synthesis filter 318, and a synthesized signal is output. The synthesis filter 318 has an inverse filter relationship with the prediction filter 314. The error between the input audio signal and the composite signal, which is the output of the subtraction circuit 319, is transformed into its spectrum by an auditory weighting filter 320, and then converted to a squared error calculation circuit 32.
1. The auditory weighting filter 320 is a filter whose transfer function is expressed as (0≦r≦1), and the weighting in the conventional example is to utilize the auditory masking effect similarly to the filter, and is detailed in Reference 2. Since this has already been done, the explanation will be omitted.

The square error calculation circuit 321 calculates the sum of squares of the perceptually weighted error signals for each code vector stored in the codebook 324 and for each phase of the drive pulse output from the phase search circuit 322. The results are output to the phase search circuit 322 and amplitude search circuit 323. The amplitude search circuit 323 searches the codebook 324 for the index of the code word that minimizes the sum of squares of the error signal for each phase of the drive pulse output from the phase search circuit 322, and searches for the index of the code word that minimizes the sum of squares of the error signal. The value is output to the phase search circuit 322, and the index of the code word that minimizes the sum of squares is held. The phase search circuit 322 manually uses the information of the selected density pattern to set the phase K of the drive pulse train to 1.
≦K≦llI D , the value is given to the drive signal generation circuit 317, and the minimum value of the sum of squares of the error signals determined for each of the D phases is received from the amplitude detection circuit 323. , outputs the phase corresponding to the smallest sum of squares among the D11 minimum values to the multiplexer 325, and at the same time notifies the amplitude search circuit 323 of the phase. The amplitude search circuit 323 outputs the index of the code word corresponding to the phase to the multiplexer 325. A multiplexer 325 multiplexes the sign of the prediction parameters, density pattern, gain, phase and amplitude of the drive pulse;
It is output to the transmission line via the output terminal 326.

Note that the output of the subtraction circuit 319 is applied to an auditory weighting filter 320.
It may be input directly to the squared error calculation circuit 321 without going through.

Next, the decoding device shown in FIG. 5 will be explained. Fifth
In the figure, the demultiplexer 331 has an input terminal 330
The code input from the prediction parameter, density pattern,
Separates the sign of the gain, phase and amplitude of the drive pulse. The decoding circuit 332337 decodes the density pattern of the driving pulse and the sign of the gain of the driving pulse, respectively, and outputs them to the driving signal generation circuit 333. The codebook 335 is the same as the codebook 324 in the encoding device shown in FIG. 4, and outputs a code word corresponding to the amplitude index of the applied drive pulse to the drive signal generation circuit 333. The prediction parameter decoding circuit 336 decodes the code of the prediction parameter encoded by the prediction parameter encoding circuit 313 in FIG. 3
Output to 46. Similar to the drive signal generation circuit 317 in the encoding device, the drive signal generation circuit 333 generates variable density or variable drive in subframe units based on the normalized amplitude of the input drive pulse and the phase of the drive pulse. generate a signal,
It is output to the synthesis filter 334 and output terminal 345. A synthesis filter 334 is a synthesis filter 3 in the encoding device.
18, which receives the drive signal and prediction parameters and outputs a composite signal to the buffer 338.

The buffer 338 combines the manually input signals frame by frame and outputs the combined signal to the output terminal 339.

[Effects of the Invention] As explained above, according to the present invention, when an uncorrectable error that exceeds the error correction capability occurs in one encoded signal sequence, it is possible to simply limit the output level. Instead, the audio signal is reproduced using the error-corrected frame synthesis filter parameters and the drive signal, so even in poor environments where the error rate of the transmission path is high, such as in mobile communications, there is little noise and no sound. You can get natural sound without any interruptions. Furthermore, in the present invention, since the interpolation process is performed for frames whose errors cannot be corrected at the level of the drive signal, even if a discontinuous point occurs in the waveform of the drive signal due to the interpolation process, it will not pass through the synthesis filter. This has the effect that the discontinuous points are smoothed out and the discontinuous sounds become indiscernible in the reproduced audio.

[Brief explanation of drawings]

FIG. 1 is a block diagram of an encoding device according to an embodiment of the present invention, FIG. 2 is a block diagram of a decoding device according to an embodiment of the present invention, and FIG. 3 is a block diagram of an encoding device according to an embodiment of the present invention. FIG. 4 is a block diagram showing an example of one side of the encoding section in FIG. 1, and FIG. 5 is a frame configuration diagram for explaining the operation of the decoding section in FIG. 6 is a block diagram showing the configuration of the prediction filter in FIG. 4, and FIG. 7 is a block diagram of a decoding device including a conventional error compensation method. 20... Encoding section, 30... Error correction encoding section, 3
0.110... Buffer, 50... Interleaving section, 120... Deinterleaving section, 130... Error correction section, 140... Error detection section, 150... Decoding section,
180... Parameter memory, 170... Drive signal memory, 180... Cutout circuit, 190... Drive signal generation circuit, 200... Synthesis filter, 210... Switch circuit, 312... Prediction parameter calculation Circuit, 3H... Misfortune circuit, 314... Prediction filter, 3
15... Density pattern selection circuit, 31G... Encoding circuit, 317... Drive signal generation circuit, 318... Gain calculation circuit, 319... Subtraction circuit, 320...'e electric shock filter , 321... Squared error calculation circuit, 322... Phase search circuit, 323... Amplitude search circuit, 324... Code blank, 325... Multiplexer, 328... Encoding circuit, 331...・Demultiplexer, 332.337.338...Decoding circuit, 3
3 34 51 52 02 04 06 ... Drive signal generation circuit, ... Synthesis filter, 338 ... Buffer, ... Long-term prediction filter, ... Short-time 1,000M filter, ... Error correction Section, 503...Decoder,...Frame period detection section,...Error rate measurement section, 507...Variable amplitude limiter,
j times

Claims (1)

[Claims] means for inputting a coded signal sequence coded for each fixed-length or variable-length frame for each frame; error correction means for correcting errors in the coded signal sequence; and error detection in the coded signal sequence. a signal means for decoding the error-corrected encoded signal sequence, a means for accumulating parameters of a synthesis filter obtained by the decoding means, and a drive signal for the synthesis filter obtained by the decoding means. means for obtaining a pitch period based on the decoded signal obtained from the decoding means; and a drive signal accumulated in the means for accumulating the drive signal when the error detection means detects an error of a certain amount or more. means for cutting out a driving signal corresponding to a pitch period from a pitch period, means for generating a driving signal for one frame using the cut out driving signal, a synthesis filter inputting this driving signal and reproducing a synthesized audio signal; An error compensation system characterized by comprising means for outputting an audio signal decoded by the decoding means when the error detection means does not detect an error of a certain value or more.