JPH07143075A - Voice coding communication system and device therefor - Google Patents

Abstract

PURPOSE:To prevent degradation of reproduced sound by using the coefficient and gain of a synthetic filter by replacing them by the values corresponding to one vector just before and performing decoding without performing a post filter processing when a flag showing an error does not exist. CONSTITUTION:The vector (b) that a exciting VQ code book 1 outputs is made into vectors (c) and (f) by the gain adjuster 2 of gain (e) and a synthetic filter 4 and the vectors are outputted to backward gain and synthetic filter adapters 3 and 5. By the gain (d) and the synthetic filter coefficient (g) that the adapters 3 and 5 predict and calculate, gain and synthetic filter coefficient control circuits 8 and 9 control the adjuster 2 and the filter 4. The circuit 9 holds or updates the coefficient (g) based on the error detection flag (k) that a radio line control processor 10 outputs and makes the filter 4 correspond to the next input vector by an output (h). After the vector (f) is processed by a post filter 6 based on the flag (k), the vector (f) is outputted as a PCM output (j) by a PCM conversion circuit 7. Thus, degradation of reproduced sound quality is prevented and an effective error countermeasure is taken.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a voice coding communication system and its apparatus, and more particularly to a low delay code excited linear prediction (Low Delay-Code Excited L) as a coding system.
The present invention relates to a speech coding communication system and a device thereof to which a 16 kbit / s speech decoding system using Inear Prediction (hereinafter referred to as LD-CELP) is applied.

[0002]

2. Description of the Related Art With the recent diversification of social and economic activities and internationalization, expectations for mobile communication are rapidly increasing.
Above all, the demand for portable car phones and cordless phones has been increasing dramatically. In addition, the third is a public land mobile communication system integrated for the 21st century.
Generation Mobile Communication System (FPLMTS: Future PublicL)
and Mobile Telecommuication System) construction is CC
Considered by IR. Along with the development flow of such public land mobile communication systems, digitization of car telephone systems and cordless telephone systems is currently in progress, and they have reached the stage of commercialization. Regarding the digital mobile phone system, in order to effectively use the frequency, the development of a half-rate system that can increase the number of channels that can be transmitted to twice the full rate without changing the transmission rate in the wireless section is underway. And its specifications are solidifying. In the near future, half-rate conversion will be promoted in digital cordless telephone systems for the same reason as in digital car telephone systems. The standard voice coding system used in digital cordless telephone systems is 32 kbit / s adaptive differential P in full rate systems.
CM (ADPCM: Adaptive Differential Pulse Code)
Modulation) has already been adopted, and in half-rate systems, low delay code excitation linear prediction (Low Delay-Code E
xcited Linear Prediction (hereinafter referred to as LD-CPEL) is expected to be used in a 16 kbit / s speech coding system.

One of the general characteristics of the transmission line of land mobile communication is that the propagation state changes abruptly and errors occur in bursts. When such a transmission error frequently occurs, the quality deterioration of the reproduced voice becomes noticeable. As a countermeasure against this, in a digital car telephone system, FEC (forward error correction) and interleaving are used, and a system resistant to transmission errors is constructed. In the cordless telephone system, the error correction code and deep interleaving cannot be used because the delay time from terminal to terminal is required to be short and the addition of redundant bits is not easy. Therefore, measures other than error correction are necessary in order to reduce the auditory deterioration due to an error in the reproduced voice after synthesis. When a transmission error occurs, the level of the reproduced voice on the receiving side often increases rapidly, and the audible quality remarkably deteriorates. In order to avoid this phenomenon by using measures other than error correction, the level of the section affected by the transmission error in the reproduced voice must be
It is considered that the method of approximating the spectral characteristics to those of the reproduced speech just before the transmission error occurs is effective in not deteriorating the perceptual quality.

[0004]

At present, with respect to the LD-CELP, which is the standard voice coding method for a half rate system of a digital cordless telephone, transmission error countermeasure techniques other than error correction codes that can be incorporated into the LD-CELP are being studied. Absent. The object of the present invention is LD-CELP.
In a digital cordless telephone system using code decoding, an error correction code and an error countermeasure function that does not use deep interleaving are provided to reduce the perceptual deterioration due to a transmission error of a reproduced voice synthesized on the receiving side. CEL
This is a method of adding to the P decoder. By adding this function, the level and spectrum characteristics of the reproduced voice in the section affected by the transmission error are approximated to those of the reproduced voice immediately before the transmission error occurs, and the deterioration of the perceptual quality is noticeable. Disappear. An object of the present invention is to carry out digital cordless telephone communication using LD-CELP coding / decoding, so that it has an error correction code and an error countermeasure function that does not require deep interleaving, and reproduces the reproduced voice to be synthesized on the receiving side. It is an object of the present invention to provide a voice coding communication system and an apparatus for the same, in which the auditory deterioration due to a transmission error is reduced.

[0005]

According to the voice coding communication system of the present invention, in digital cordless telephone communication, in order to reduce the auditory deterioration due to transmission error of reproduced voice synthesized at the receiving side, the transmitting side The voice signal is coded and transmitted by the LD-CELP encoder for each vector with 5 samples as one vector at the sampling frequency of 8 kHz, and on the receiving side, the received signal is decoded by the LD-CELP decoder for each vector. In order to reduce the effect of transmission errors on reproduced voice in a voice coding communication system in which a reproduced signal is output to, the line quality is constantly monitored by a wireless line control processor, a transmission error is detected, and an error detection flag is output. On the receiving side, the error detection flag output from the wireless line control processor is received in the section indicating that there is no error. The signal is decoded by the LD-CELP decoder to output a reproduced signal, and the LD-CELP is used in a section in which the error detection flag indicates that there is an error.
The coefficient of the synthesis filter, which is an internal parameter of the decoder, and the gain of the gain adjuster are used by replacing them with the values corresponding to one vector immediately before the error occurs, and the post filter processing is controlled not to be performed. It is characterized in that the reproduced signal is decoded by the above.

A speech decoding apparatus of the present invention for carrying out such a communication system is such that an input speech signal is L vector by vector.
A D-CELP encoded signal is received via a transmission line, and LD-CELP is performed while reducing the influence of a transmission error.
In a voice decoding device that decodes a received signal by decoding and reproduces a reproduced voice for each vector, a radio line control processor that constantly monitors line quality, detects a transmission error, and outputs an error detection flag indicating the presence or absence of a transmission error. If there is no error based on the error detection flag output from the radio line control processor, the received signal for each vector is input and decoded, and the reproduced signal is output for each vector, and the error is detected. In this case, the coefficient of the synthesis filter, which is an internal parameter, and the gain of the gain adjuster are used by replacing them with the value corresponding to one vector immediately before the error occurs, and the post-filter processing is not performed. Decode the signal 1
An LD-CELP decoder that outputs reproduced voice for each vector is provided.

Further, in the speech decoding apparatus, when the error detection flag is switched from "with transmission error" to "without transmission error", the gain value of the gain adjuster of the LD-CELP decoder is immediately before the occurrence of the error. When the value that has been replaced / fixed with the value corresponding to one vector of is updated to the latest gain value calculated by the backward gain adaptor, the [Use for decoding the current vector] Gain = 1 The gain used in decoding before vector × α + the latest gain calculated by the backward gain adaptor × (1−α)] to smooth the variation of the gain to eliminate the abnormal noise generated in the reproduced voice. The feature is that the amount is reduced.

[0008]

1 is a block diagram of an LD-CELP decoder according to the present invention. In the figure, the wireless line control processor 1
0, the gain control circuit 8, and the synthesis filter coefficient control circuit 9 are
It is a circuit added by the present invention. The other part is LD-CELP shown in the standard TTC JT-G728.
It is a component that operates exactly the same as the decoder. The radio circuit control processor 10 constantly monitors the line quality, detects a transmission error, and indicates an error detection flag k indicating the presence or absence of the transmission error.
Is output. Here, CRC is used as an error detection method.
(Cyclic redundancy code: publicly known technique). The input signal a is one vector (5
This is an index expressing the shape and amplitude of the optimum excitation vector b for the sample), and the excitation VQ (Vector Quantization) is calculated using this input signal a.
The optimum excitation vector b is selected from the codebook 1 and output. The excitation vector b is gain-adjusted by the gain adjuster 2 using the gain e to obtain the gain-adjusted excitation vector c. Here, the backward gain adaptor 3 calculates the updated gain d for performing the gain adjustment for the next input vector by backward prediction from the past gain adjusted excitation vector c. The gain control circuit 8 fixes (holds) the past gain d based on the error detection flag k from the wireless line control processor 10, or passes (updates) the latest gain d as it is, or performs a smoothing process to the gain adjuster 2 The gain e used in is determined and output (this detailed operation,
Timing will be explained later). Gain control circuit 8
The output e of is used by the gain adjuster 2 as the gain corresponding to the next input vector.

Next, the synthesis filter 4 is driven by the gain adjusted excitation vector c to synthesize the reproduced voice vector f. Here, the backward synthesis filter adaptive unit 5 calculates and outputs the synthesis filter coefficient g corresponding to the next input vector by backward prediction from the past reproduced voice vector f. The synthesis filter coefficient control circuit 9 holds or passes (updates) the synthesis filter coefficient g as it is based on the error detection flag k from the wireless line control processor 10 (this timing will be described later). The output h of the synthesis filter coefficient control circuit 9 is used by the synthesis filter 4 as a synthesis filter coefficient corresponding to the next input vector. The reproduced voice vector f is processed by the post filter 6 to improve the quality in hearing, and then output by the PCM conversion circuit 7 as a 64 kbit / s μ-law PCM or 16-bit uniform PCM output j. The post filter 6 has a function of entering a sleep mode based on the error detection flag k.

Next, the operations of the gain control circuit 8, the synthesis filter coefficient control circuit 9 and the post filter 6 will be described with reference to the time chart of FIG. (A) is LD-CEL
3 shows a frame of an audio signal input to a P encoder. The description will proceed on the assumption that one frame length is 10 msec and each frame is composed of 16 vectors (80 samples; sampling frequency 8 kHz). The voice code data string by the LD-CELP encoder corresponding to this frame structure is input to the LD-CELP decoder on the receiving side. (B)
Is an error detection flag k determined by the wireless link control processor 10 on the receiving side, and indicates whether or not a transmission error has occurred in frame units.
Detect nothing. Here, a transmission error has occurred in the transmission data bit string of frame 2. (C) is LD-CELP
The operation of the gain control circuit 8 in the decoder is shown. When the error detection flag k indicates no error (frames 1, 3, 4)
Is for gain update (passing input as it is, output) or gain smoothing processing, and if there is an error (frame 2)
Holds and outputs the gain corresponding to the last 1 vector of the frame (here, frame 1) immediately before the error occurs (detailed control operation will be described later). By this processing, the level of the reproduced voice in the section affected by the transmission error can be approximated to that of the reproduced voice immediately before the transmission error occurs.

(D) shows the operation of the synthesis filter coefficient control circuit 9 in the LD-CELP decoder. Error detection flag k
Indicates no error (frames 1, 3, 4),
When the gain is updated (passes the input as it is and is output) and there is an error (frame 2), the gain corresponding to the last 1 vector of the frame immediately before the error (here, frame 1) is held and output. (Detailed control operation will be described later). By this processing, the spectrum characteristic of the reproduced voice in the section affected by the transmission error can be approximated to that of the reproduced voice immediately before the transmission error occurs.
(E) shows the operation of the post filter 6, and when the error detection flag k indicates that there is no error (frame 1,
3 and 4) operate normally, and when there is an error (frame 2), the operation is stopped and the input signal is output as it is without being processed. The reason for suspending the Bost filter when an error occurs (passing the input as it is without filtering) is to perform spectral envelope enhancement and pitch enhancement in order to improve the perceptual quality of the reproduced voice. This is because, if these processes are performed on the reproduced voice affected by the transmission error, it may have an adverse effect and the influence of the transmission error may be emphasized.

Next, the operation of the gain control circuit 8 will be described in detail with reference to FIG. The flowchart shown here is executed for each frame. ~ Indicates a step number. : Check whether the transmission error detection flag k indicates that there is an error. : When there is a transmission error, the gain d corresponding to the last one vector of the frame immediately before the transmission error occurs is held, and the held value is maintained while the transmission error continues to occur. Output e. : When the succeeding frame has no transmission error, the gain smoothing process is performed, and therefore, a specific numerical value M is set in the control counter G as preparation for the gain smoothing process. Here, M is the number of frames for which gain smoothing is continued (for example, 3). : If there is no error, check whether the value of the control counter G is 0. : If the value of the control counter G is 0, the gain smoothing process is not performed, and the output d from the backward gain adaptor 3 is passed as it is as the output e. : If the value of the control counter G is not 0, the gain smoothing process is continued in this frame. Subtract 1 from the control counter G. : Performs gain smoothing processing. At the time when the error detection flag is switched from “with transmission error” to “without transmission error”, the value of the gain e used in the gain adjuster 2 corresponds to the state (1 vector (5 samples) immediately before the error occurrence). State (value replaced / fixed with a value) (the latest gain value d calculated by the backward gain adaptor 3 is defined as e)
When it is instantly updated to, the gain value e used for decoding changes abruptly, which may cause an abnormal sound in the reproduced voice. To avoid this, a predetermined time (M
(Specified by) only performs gain smoothing processing as shown below. The gain e used by the gain adjuster 2 is e = 1 vector The gain used before e × α + the latest gain calculated by the backward gain adaptor d × (1−α) where, for example, α = 0.95. . As described above, the change in the gain is smoothed at the time of switching from the transmission error to the transmission error-free, thereby reducing the abnormal noise generated in the reproduced voice.

FIG. 3 shows the operation of the synthesis filter coefficient control circuit 9. The flowchart shown here is executed for each frame. : Check whether or not the transmission error detection flag k indicates that there is an error. : When there is a transmission error, the synthesis filter coefficient g corresponding to the last one vector of the frame immediately before the transmission error occurs is held, and is held while the transmission error continues to occur. The value is output h. : When there is no transmission error, the output g from the backward synthesis filter adaptive unit 5 is passed as it is as the output h. Although the constituent elements of the circuit of the present invention have been described as to how to implement the circuit, the LD-CELP encoder, LD-C
Since the ELP decoder is generally realized by a signal processing microprocessor (DSP),
The peripheral circuits of the LD-CELP codec described here can be easily realized as a part of the program of the DSP.

[0014]

EFFECT OF THE INVENTION Half-ray of digital cordless telephone
In the audio system, by carrying out the present invention, the level of the reproduced voice in the section affected by the transmission error,
The spectral characteristics are approximated to those of the reproduced voice just before the transmission error occurs, and the audible quality deterioration becomes inconspicuous. Therefore, it is possible to provide a speech decoding communication system having an effective error countermeasure function that does not require an error correction code and deep interleaving, and an apparatus thereof.

[Brief description of drawings]

FIG. 1 is a configuration diagram showing an embodiment of the present invention.

FIG. 2 is a flowchart illustrating the operation of the gain control circuit of the present invention.

FIG. 3 is a flowchart illustrating the operation of the synthesis filter coefficient control circuit of the present invention.

FIG. 4 is a timing chart showing the operation of the present invention.

[Explanation of symbols]

 1 Excitation VQ Codebook 2 Gain Adjuster 3 Backward Gain Adaptor 4 Synthesis Filter 5 Backward Synthesis Filter Adaptor 6 Post Filter 7 PCM Conversion Circuit 8 Gain Control Circuit 9 Synthesis Filter Coefficient Control Circuit 10 Radio Channel Control Processor

Claims (3)

[Claims] 1. A transmitter side encodes a voice signal at a sampling frequency of 8 kHz by an LD-CELP encoder for each vector having 5 samples as one vector, and transmits the encoded voice signal. At a receiver side, the received signal is LD-CELP. In a voice coding communication system in which a decoder decodes and outputs a reproduction signal for each vector, and a radio line control processor constantly monitors the line quality to detect a transmission error and output an error detection flag. In order to reduce the effect of transmission error, on the receiving side, the received signal is decoded by the LD-CELP decoder in a section in which the error detection flag output from the wireless line control processor indicates that there is no error, and the reproduced signal is reproduced. In the interval indicating that the error detection flag has an error, the internal parameter of the LD-CELP decoder is output. The synthesized filter coefficient and the gain of the gain adjuster are replaced with the values corresponding to the one vector immediately before the error occurs, and the post-filter processing is controlled so that the reproduced signal is A voice coding communication method characterized by being decoded. 2. An LD-C input speech signal for each vector
A voice decoding device for receiving an ELP-encoded signal via a transmission line, decoding the received signal by an LD-CELP decoder while reducing the influence of a transmission error, and reproducing reproduced voice for each vector, When there is no error based on the wireless line control processor that constantly monitors the quality, detects the transmission error, and outputs an error detection flag indicating the presence or absence of the transmission error, and the error detection flag output from the wireless line control processor Receives the received signal for each vector, decodes it and outputs the reproduced signal for each vector. When an error is detected, an error occurs in the coefficient of the synthesis filter and the gain of the gain adjuster which are internal parameters. Replace with the value corresponding to the previous 1 vector and use it,
LD-CEL that decodes the received signal without performing post-filter processing and outputs reproduced sound for each vector
A speech decoding device comprising a P decoder. 3. The speech decoding device according to claim 2, wherein
At the time when the error detection flag switches from “with transmission error” to “without transmission error”, the gain value of the gain adjuster of the LD-CELP decoder is replaced / fixed with the value corresponding to one vector immediately before the error occurrence. Value is updated to the latest gain value calculated by the backward gain adaptor, for a predetermined time, [gain used for decoding current vector = 1 gain used for decoding before vector] × α + latest gain calculated by the backward gain adaptor × (1-α)]
3. The speech decoding apparatus according to claim 2, wherein abnormal noise generated in the reproduced speech is reduced by smoothing the change in gain.