JPH0637725A - Radio communication equipment - Google Patents

Abstract

PURPOSE:To provide a radio communication equipment in which voice quality after error frame processing is improved by taking production of a listening sense of a peak noise after error processing of an error frame into account. CONSTITUTION:An attenuation level is varied with the number of consecutive error frames and attenuation is applied even to a normal frame after the error frames for a prescribed period in a level discrimination block 12. A fluctuation of an output level is checked based on a scale factor and a level of an output waveform when the scale factor exceeds a threshold level within a prescribed frame period for suppression is suppressed in a peak noise suppression block 8.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a wireless communication device applicable to a digital cordless telephone or the like.

[0002]

2. Description of the Related Art A voice codec used in a second-generation cordless telephone, which is currently being standardized, is a CCI.
TT Recommendation G. 721 32 kbps ADPCM is to be used. G.C. which is recommended by CCITT.
The standard 721 is originally made on the assumption that it is used for a wired path having a low error rate, and is not considered for use in a wireless path where various errors may occur. Therefore, when used in a second-generation cordless (digital cordless) telephone, sufficient communication quality cannot be obtained unless some kind of error countermeasure is taken. Therefore, each company is researching the countermeasure.

In a digital cordless telephone, ADPCM data which is a voice signal has a CRC code (CCIT) for error detection with 40 samples as one frame.
16 bits are used according to T recommendation), and TDMA is added.
It is transmitted as a (time division multiple access) method. In a digital cordless telephone, 40 samples (5 milliseconds) are set as one frame and are transmitted by time division for each frame. Then, when there is a transmission error in the ADPCM data in this frame, the receiving side recognizes it as a CRC error and performs a predetermined process.

Various studies and presentations have been made to improve the quality of reproduced voice in this type of digital cordless telephone. Here, as one of the conventional techniques related to digital cordless telephones, a method proposed by NTT will be mentioned (Tanaka, Kobayashi, Suzuki, Takagi, “Voice quality improvement method for digital cordless telephones” 1
991 IEICE Spring National Convention B-414,
Tanaka, Hamada, Hirono, "A study of digital cordless telephone system", 1992 IEICE Spring National Convention B-329).

That is, in the above method, it is considered that the burst quality is the main cause of the voice quality deterioration in the digital cordless telephone, and as a countermeasure against the transmission error, the error frame is processed by the following two methods (see FIG. 7).
Muting with.

(1) Among the data of the frame having an error, the data exceeding a certain threshold value is replaced with the threshold value in the data replacing section 70.

(2) A certain amount of attenuation is applied by the attenuator 72 at the analog output point of the frame having the error.

[0008]

However, while the above-mentioned conventional technique is effective in suppressing the peak noise of the burst noise, the ADPCM data in the frame subjected to the error processing and the ADPCM data of the subsequent normal frame are effective. Noise may occur due to discontinuous levels.

Further, when there are few errors in one frame such as a random error, if an error countermeasure is taken, a large peak noise may occur in the subsequent normal frame. In this case, if the error countermeasure is taken, the voice quality will be deteriorated. More specifically, it is as follows.

That is, ADP in which a transmission error has occurred
Since the CM data is replaced with a predetermined value by the data replacement unit 70 (see FIG. 7), the decoded output obtained from the ADPCM decoder 71 has a certain suppressed low level. However, if a large level is decoded in a normal frame following the error frame, the audio level is suddenly changed from a low audio level to a high audio level, so that the same audibility as that containing peak noise is obtained. by.

It is therefore an object of the present invention to provide a wireless communication apparatus with improved voice quality after error frame processing in view of the possibility that peak noise-like sensation may occur after error frame error countermeasures.

[0012]

In order to achieve the above object, according to the present invention, as shown in FIG. 8, a transmission error occurs in ADPCM data of a predetermined frame and ADPCM data of a predetermined bit as one frame. And an error signal indicating that the ADPCM data has a transmission error in the frame, the data replacement means 80 that replaces the ADPCM data with a predetermined value so that the ADPCM data has a predetermined threshold value or less. And an AD for decoding the output signal from the data replacing means.
The ADPCM corresponding to the frame in which the transmission error has occurred by inputting the output signal from the PCM decoder means 81 and the ADPCM decoder means and the error signal.
Attenuation processing means 82 is provided for performing attenuation processing on a signal from the decoder means at a predetermined level and performing attenuation processing on a predetermined number of normal frames following the error frame at a predetermined level.

Further, in another embodiment of the present invention, as shown in FIG. 9, the ADPCM data and an error signal indicating that a transmission error has occurred in the ADPCM data of the frame with the ADPCM data of a predetermined bit as one frame. From the data replacement means 90, which replaces the ADPCM data with a predetermined value so that the ADPCM data becomes equal to or less than a predetermined threshold when a transmission error occurs in the frame after inputting. ADP which decodes the output signal of and outputs a scale factor indicating the adaptive speed of the decoded signal
For the CM decoder means 91 and a predetermined number of normal frames following the error frame, the signal from the ADPCM decoder means is suppressed to a predetermined threshold value or less when the scale factor is a predetermined threshold value or more. Suppressing means 92 for controlling

[0014]

According to the present invention shown in FIG. 8, in the attenuation processing means 82, the attenuation level is changed in accordance with the number of consecutive error frames, and attenuation is performed in a certain section even for a normal frame after the error frame. Try to add.

Further, in the present invention shown in FIG. 9, the suppressing means 92 suppresses a point where the output level largely changes within a constant frame after the error frame. That is, the fluctuation of the output level depends on the scale factor obtained from the ADPCM decoder 91 (since it is a parameter indicating the adaptive speed, the magnitude of the level fluctuation (that is, the presence of a peak) can be known from the magnitude of this scale factor). By checking, the level of the output waveform at a point where the scale factor exceeds a certain threshold value is suppressed within a predetermined frame section where suppression is performed. As a result, the peak noise after the error frame is suppressed.

[0016]

EXAMPLES Examples of the present invention will be described in detail below.

FIG. 1 is a block diagram showing an entire embodiment of the present invention. In the figure, 2 is a data replacement block, which is the ADPCM extracted from the TDMA frame.
The received data (32 kbps) and the error signal obtained from the CRC code are input, and data above a predetermined threshold value is replaced with the threshold value.

Reference numeral 4 is an ADPCM decoder compliant with the digital cordless telephone standard (CCITT G.721). Similarly, 5 is CCITT G. 711-compliant PCM decoder. The scale factor (y) output from the ADPCM decoder 4 is
A to indicate the adaptive speed (variation) of the decoded signal
This is a well-known parameter used in the DPCM decoder, but in the present embodiment, the peak noise suppression process described later is performed based on this scale factor.

Reference numeral 8 is a peak noise suppression block, 10 is an attenuator, 12 is a level determination block for setting an attenuation level according to the number of error frames, and 14 is a speaker.

FIG. 2 is a flow chart showing the operation of the data replacement block 2. Step S21 in this figure
First, the ADPCM received data and the CRC error signal are input. In the next step S22, the presence or absence of a transmission error is determined based on the error signal.

If no transmission error has occurred, the received data is output as it is in step S23.
On the other hand, if a transmission error has occurred, step S2
4, the absolute value of the received data is a predetermined "threshold value"
If it is larger than the threshold, and if a positive determination is obtained, then in step S25, the received data is replaced with a threshold value and output. If a negative determination is obtained in step S24, the received data is output as it is (step S23).

The ADPCM decoder 4 and the PCM decoder 6 shown in FIG. 1 are CCIT as described above.
Since it is a standard that complies with T's recommendation, description thereof is omitted here.

FIG. 3 is a circuit diagram showing the peak noise suppression block 8 in detail. In the figure, reference numeral 31 is a suppression section determination block, which determines whether or not a frame section in which suppression should be performed, based on the input error signal. Then, when it is determined that the section is to be suppressed, the changeover switch 32 is tilted to the A side to perform the suppression processing described later, while when it is determined that the section is the non-suppressed section, the switch 32 is tilted to the B side. The input audio signal is output as it is. Here, the section to be suppressed includes not only the error frame but also several normal frame sections following the error frame.

It is also possible to set only the normal error frame section following the error frame as the suppression section and not perform the suppression processing on the error frame itself (that is, the switch 32 is tilted to the B side).

Reference numeral 33 is a suppression sample judgment block,
When the suppression section determination block 31 determines that the suppression section is a suppression section, the adaptive speed (variation) of the decoded signal
It is determined whether or not the current voice data should be suppressed by using a scale factor that represents. That is, when the scale factor exceeds a predetermined threshold value, the changeover switch 34 is tilted to the B side in order to suppress the audio data. If the scale factor is equal to or less than the threshold value even within the same frame section, the switch 34 is tilted to the A side and the suppression process is not performed.

Reference numeral 35 is a signal suppressing block, and 36 is a buffer, and both of these blocks perform an operation for suppressing processing shown in the lower part of FIG. The suppression process here is
In order to reduce the rate of change (difference) between the previous data and the current data, a buffer 36 for holding the output data is provided.

FIG. 4 is a flow chart when the peak noise suppression processing shown in FIG. 3 is realized by a DSP (digital signal processor). However, in this flowchart, the suppression process is performed only on the data of the two frames following the error frame, and the peak noise suppression process is not performed on the error frame.

When the peak noise suppressing process is executed by the DSP, the current data level is lowered as the suppressing process in the flowchart shown in FIG.
No buffer equivalent to the buffer 36 shown in FIG. 3 is required.

In step S41 of FIG. 4, voice data, a squeeze factor, and an error signal are input.

In the next step S42, it is determined whether or not the data is for two frames following the error frame.
As a result, when a negative determination is obtained, it means that the data is not within the two frames after the error frame, and therefore the received data is output as it is in step S43.

When an affirmative determination is obtained in step S42, the process proceeds to step S44, and it is determined whether the skate factor is larger than the threshold value. As a result, when a negative determination that the scale factor is not larger than the threshold value is obtained, the reception data is output as it is (step S43), while an affirmative determination that the scale factor is larger than the threshold value is obtained. If so, step S
At 45, the received data is suppressed by the suppression coefficient sup and output.

FIG. 5 is a state transition diagram showing a judgment method in the level judgment block 12 shown in FIG. The numbers in the circles and squares in this figure indicate the respective states, the numbers on the arrows indicate the conditions for the state transition, 0 indicates no error, and 1 indicates the error. That is, the states 0, 4, 5, and 6 represent the processing on the data in the normal frame, and the states 1, 2, and 3 represent the processing on the data in the error frame. Further, each one square represents the length of one frame. Settle, in this example,
The states 6, 5, and 4 mean that they continue for 3, 2, and 1 frame, respectively. For states 1, 2, and 3,
For example, it is assumed that 10 frames are consecutive.

Then, as shown in FIG. 6, a certain attenuation value is assigned to each state. That is, as a whole, the current state is known from the input error signal, and the attenuation value assigned to that state is output.

[0034]

As described above, according to the present invention, the audio output stage is provided with the attenuation processing means which operates even after the error frame, and the suppression means for suppressing the peak noise after the error frame is added. Therefore, the special effects listed below can be obtained.

(1) At the time of a burst error, the noise after the error frame, which was heard in the conventional error countermeasure method, is suppressed and the voice quality is improved. Furthermore, the peak noise that sometimes occurs is suppressed.

(2) In the case of noise with a low error rate,
The peak noise caused by the conventional error countermeasure is
It is suppressed to the extent that the difference in hearing is not noticeable.

(3) According to the present invention, it is possible to take an error countermeasure against both noises without judging whether the noises are random noises or burst noises.

[Brief description of drawings]

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

FIG. 2 is a flowchart showing the operation of the data replacement block shown in FIG.

3 is a block diagram showing in detail the configuration of a peak noise suppression block shown in FIG.

FIG. 4 is a flowchart when performing peak noise suppression processing using a digital signal processor.

5 is a state transition diagram showing the operation of the level determination block shown in FIG. 1. FIG.

FIG. 6 is a diagram showing operations of a level determination block and an attenuator shown in FIG. 1.

FIG. 7 is an explanatory diagram of a conventional technique.

FIG. 8 is a schematic configuration diagram of the present invention.

FIG. 9 is another schematic configuration diagram of the present invention.

[Explanation of symbols]

 2 Data Replacement Block 4 ADPCM Decoder 6 PCM Decoder 8 Peak Noise Suppression Block 10 Attenuator 12 Level Judgment Block 14 Speaker 31 Suppression Section Judgment Block 32 Changeover Switch 33 Suppression Sample Judgment Block 34 Changeover Switch 35 Signal Suppression Block 36 Buffer

Claims (3)

[Claims] 1. ADPCM data and AD of predetermined bits
ADP of PCM data as one frame
An error signal indicating that a transmission error has occurred in the CM data is input, and when the transmission error occurs in the frame, the ADPCM data is set to a predetermined threshold value so that the ADPCM data becomes equal to or less than a predetermined threshold value. A data replacement unit that replaces a value, an ADPCM decoder unit that decodes an output signal from the data replacement unit, an output signal from the ADPCM decoder unit, and the error signal are input to a frame in which a transmission error occurs. Attenuation processing means for performing attenuation processing at a predetermined level on a signal from the corresponding ADPCM decoder means and for performing attenuation processing at a predetermined level on a predetermined number of normal frames following the error frame. A characteristic wireless communication device. 2. The wireless communication device according to claim 1, wherein the attenuation processing means changes and sets the attenuation level according to the number of consecutive error frames. 3. ADPCM data and AD of predetermined bits
ADP of PCM data as one frame
An error signal indicating that a transmission error has occurred in the CM data is input, and when the transmission error occurs in the frame, the ADPCM data is set so that the ADPCM data becomes equal to or less than a predetermined threshold value. Data replacement means for replacing with a predetermined value, ADPCM decoder means for decoding the output signal from the data replacement means, and outputting a scale factor indicating the adaptive speed of the decoded signal, and a predetermined number following the error frame. The normal frame, the suppressing means suppresses the signal from the ADPCM decoder means to a predetermined threshold value or less when the scale factor is a predetermined threshold value or more. Communication device.