JP3436940B2 - Wireless communication device - Google Patents

Description

Description: BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a radio communication device applicable to digital cordless telephones and the like. 2. Description of the Related Art A voice codec used in a second generation cordless telephone which is currently being standardized is CCI.
TT Recommendation G. 721 32 kbps ADPCM is to be used. G. CC recommended by CCITT.
The 721 standard is originally created on the assumption that it is used on a wired path having a low error rate, and is not considered for use on a wireless path where various errors may occur. For this reason, when used for a second-generation cordless (digital cordless) telephone, sufficient communication quality cannot be obtained unless some error countermeasures are taken, and companies are studying countermeasures. In a digital cordless telephone, ADPCM data, which is an audio signal, is divided into a CRC code (CCIT) for error detection using 40 samples as one frame.
T-recommendation uses 16 bits), and TDMA
It is transmitted as (time division multiple access) system. In a digital cordless telephone, 40 samples (5 milliseconds) are set as one frame and transmitted in a time-division manner 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 predetermined processing. [0004] Various studies and presentations have been made to improve the quality of reproduced sound in this kind of digital cordless telephone. Here, we refer to one of the methods proposed by NTT as one of the prior arts related to digital cordless telephones (Tanaka, Kobayashi, Suzuki, Takagi, "Method of improving voice quality of digital cordless telephones" 1.
991 IEICE Spring National Convention B-414,
Tanaka, Hamada, Hirono, "A Study of Digital Cordless Telephone System" 1992 IEICE Spring Convention B-329). [0005] That is, in the above method, it is considered that voice quality is degraded in a digital cordless telephone due to a burst error as a main factor.
Muting with (1) Among the data of the frame in which an error has occurred, data exceeding a certain threshold is replaced with a threshold by the data replacement unit 70. (2) A certain attenuation is applied by an attenuator 72 at an analog output point of a frame having an error. However, in the above-mentioned prior art, the peak noise among the burst noises is suppressed and is effective, but the ADPCM data in the error-processed frame and the ADPCM data in the subsequent normal frame are effective. The level with the data may be discontinuous and noise may occur. Further, when errors in one frame are small, such as random errors, taking measures against errors may cause a large peak noise in a subsequent normal frame. In this case, taking a countermeasure against an error will deteriorate the voice quality. More specifically, it is as follows. That is, ADPCM data in which a transmission error has occurred is replaced with a predetermined value by the data replacement unit 70 (see FIG. 8).
The decoded output obtained from the DPCM decoder 71 will exhibit a certain suppressed low level. However, if a large level is decoded in a normal frame following an 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 including peak noise is obtained. by. [0009] In order to achieve the above object, the present applicant has proposed a method in which a transmission error occurs in a frame.
ADP so that DPCM data is less than a predetermined threshold
The CM data is replaced with a predetermined value, and then the signal of the frame in which the transmission error has occurred is subjected to attenuation processing at a predetermined level, and the attenuation processing is performed at a predetermined level for a predetermined number of normal frames following the error frame. It was suggested to do. [0010] Further, the signal from the ADPCM decoder means is converted into an ADPCM signal when a scale factor indicating the adaptation speed of the decoded signal is equal to or larger than a predetermined threshold value for a predetermined number of normal frames following the error frame.
It has been proposed to suppress the signal from the M decoder means below a predetermined threshold. With such a configuration, it is possible to prevent the occurrence of noise generated in the output voice due to the discontinuity of the output voice level, and to cope with both burst errors and random errors. That is, in the above configuration, at the stage of the attenuation process, a burst error and a random error are distinguished so that an appropriate process corresponding to each error can be performed. However, replacement of ADPCM data has the disadvantage of improving sound quality for burst errors but of degrading sound quality for random errors. For this reason, there is a problem that it is not possible to sufficiently cope with a burst error and a random error only by the attenuation process. In view of the above, it is an object of the present invention to provide a radio communication apparatus capable of performing sufficient processing for both random errors and burst errors. [0013] In order to solve the above-mentioned problems, the present invention provides a method in which ADPCM data is input and the absolute value of the ADPCM data is equal to or less than a predetermined threshold value. A data replacement unit for replacing the ADPCM data with a predetermined value; and an error signal indicating that a transmission error has occurred in the ADPCM data of the frame with the predetermined bit of the ADPCM data as one frame, and an error having the error signal. Threshold setting means for changing and setting the predetermined threshold of the error frame and the subsequent normal frames according to the number of consecutive frames;
ADPCM decoder means for decoding an output signal from the data replacement means, an output signal from the ADPCM decoder means, and the error signal are inputted, and the A signal corresponding to the error frame and the subsequent normal frame is inputted.
An attenuation processing unit for performing an attenuation process on the signal from the DPCM decoder at a level determined according to the number of consecutive error signals. According to the present invention, since the threshold value for replacing the ADPCM data is changed by the error signal, the sound quality is improved with respect to the burst error. Embodiments of the present invention will be described below in detail. FIG. 1 is a block diagram showing an entire embodiment of the present invention.
In this figure, 2 is a data replacement block, and TD
ADPCM reception data (3
2 kbps) and a threshold value, and replaces the data above the input threshold value with the threshold value. 3 is a threshold value judgment block,
An error signal obtained from the C code is input, and a threshold value input to the data replacement block 2 is set. 4 is a digital cordless telephone standard (CCITT G.4).
721) is an ADPCM decoder. 5 is CCITT G. 711-compliant PCM decoder. The scale factor (y) output from the ADPCM decoder 4 is a well-known parameter used in the ADPCM decoder to indicate the adaptation speed (fluctuation) of the decoded signal. Performs a peak noise suppression process described later based on this scale factor. 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 flowchart showing the operation of the data replacement block 2. Step S in the figure
At 21, the user first inputs ADPCM reception data and a threshold value. In the next step S22, it is determined whether or not the absolute value of the received data is larger than a predetermined “threshold”,
If an affirmative determination is obtained, the received data is replaced with a threshold and output in step S24.
If a negative determination is obtained in step S22, the received data is output as it is (step S22).
23). As described above, in this embodiment, all ADPs
The determination processing of the CM data with the threshold value is performed. This can be achieved by setting the threshold value in a normal frame having no error signal to the maximum value of the received ADPCM data. FIG. 3 shows the data replacement block 2
FIG. 6 is a diagram showing an output of a threshold value determination block 3 for setting a threshold value input to the threshold value determination unit. As shown in the figure, thresholds are assigned according to error signals, and each state is indicated by a number in parentheses. For example, state 0 is a case where a normal frame continues, and is set to the maximum value of the received ADPCM data. In states 1 to 6, thresholds are set according to error signals, and the thresholds are set to 6, 4, and 2, respectively, according to the number of times an error frame appears continuously. This makes it possible to reduce the size of the output audio signal at the time of a burst error in which error frames continue. FIG. 4 shows the data replacement block 2 described above.
FIG. 6 is a state transition diagram showing the determination method in FIG. In the figure, the numbers in circles and squares indicate each state, the numbers on the arrows indicate conditions for changing the state, and 0 indicates that there is no error.
1 indicates a case where there is an error. In other words, state 0,
4, 5, and 6 represent processing for data in a normal frame, and states 1, 2, and 3 represent processing for data in an error frame. Each square represents the length of one frame. That is, in this example, the states 1, 2, 4, 5, and 6 last for 1, 2, 1, 2, and 3 frames, respectively. The ADPCM decoder 4 and the PCM decoder 6 shown in FIG.
Since the standard is based on the recommendation of T, description thereof is omitted here. FIG. 5 is a flowchart showing the operation of the peak noise suppression block 8 in detail.
It can be realized by a P (Digital Signal Processor). In the figure, first, in step S51, the audio data, error signal,
The scale factor from the PCM decoder 4 is input. In the next step S52, it is determined based on the input error signal whether or not the frame section is to be suppressed. That is, the frame is an error frame based on the error signal or 2 following the error frame.
It is determined whether the frame is an error frame or two frames following the error frame. If an affirmative determination is obtained in step S52, the process proceeds to step S53, where it is determined whether or not the current voice data should be suppressed using a scale factor representing the adaptive speed (fluctuation) of the decoded signal. . That is, it is determined whether the scale factor is equal to or less than a predetermined threshold. If the scale factor is equal to or smaller than the threshold, the audio signal from the PCM decoder 6 is output as it is (step S54). If the scale factor is larger than the threshold value, the audio signal is suppressed and output in step S55. The suppression is performed, for example, by the following arithmetic processing. ## EQU1 ## If a negative determination is obtained in step S52, the process proceeds to step S54, and the audio data is output as it is. Finally, the output data is stored for the suppression processing of the next frame (step S56). FIG. 6 is a diagram illustrating the operation of the attenuator 10. In the figure, when an error frame appears continuously, it is determined as a burst error, the attenuation level added to the audio data is lowered, and the noise generated in the output audio is suppressed. In addition, by returning the normal frame after the error frame to a staircase shape, click noise caused by rapid fluctuation of audio data is prevented. The attenuation level is controlled by the output of the level determination block 12. The attenuation level changes from state 0 to state 6 in response to an error signal, for example, as shown in the figure, and returns to state 0 again. The attenuation level is 0d in state 0, for example.
B, -3 dB for states 1 and 6, -1 for states 2 and 4
2 dB, and states 3 and 4 are -18 dB. FIG. 7 is a state transition diagram showing a determination method in the level determination block 12 shown in FIG. In the figure, the numbers in circles and squares indicate each state, the numbers on the arrows indicate the conditions for changing the state, 0 indicates no error, and 1 indicates the error. That is, states 0, 4, 5, and 6 represent processing on data in a normal frame, and states 1, 2, and 3 represent processing on data in an error frame. Each square represents the length of one frame. That is, in this example,
States 1, 2, 4, 5, and 6 are 1, 2, 1, 2, 4,
This means that it lasts for a frame. In this embodiment, the number of error frames is 1
Is a random error section, the number of consecutive error frames is 2
3 is a random-burst transition section and the number of error frames is 3.
The above is a burst error section. In addition, even after a normal frame is changed from an error frame, the state is returned through the states 4, 5, and 6, thereby preventing a rapid change in audio level. As described above, according to the present invention, the ADP
Since the threshold value for replacing the CM data is changed by the error signal, the following special effects can be obtained. (1) Peak noise in an error frame can be suppressed. (2) For a random error and a burst error, error correction processing corresponding to each error can be performed.

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram showing an overall configuration in one embodiment of the present invention. FIG. 2 is a flowchart illustrating an operation of a data replacement block illustrated in FIG. 1; FIG. 3 is a diagram illustrating an output example of a threshold determination block illustrated in FIG. 1; FIG. 4 is a diagram showing a state transition of a threshold determination block shown in FIG. 1; FIG. 5 is a flowchart showing processing of a peak noise suppression block shown in FIG. 1; FIG. 6 is a diagram illustrating operations of a level determination block and an attenuator illustrated in FIG. 1; FIG. 7 is a state transition diagram showing the operation of the level determination block shown in FIG. FIG. 8 is an explanatory diagram of a conventional technique. [Description of Signs] 2 Data replacement block 3 Threshold determination block 4 ADPCM decoder 6 PCM decoder 8 Peak noise suppression block 10 Attenuator 12 Level determination block 14 Speaker

Continuation of the front page (56) References JP-A-4-263528 (JP, A) JP-A-6-37725 (JP, A) Shigeru Tanaka, et al., B-414 Voice quality improvement method for digital cordless telephones , 1991 IEICE Spring National Convention Symposium (2), Japan, The Institute of Electronics, Information and Communication Engineers, March 15, 1991, 2-414 (58) Fields surveyed (Int. Cl. ⁷ , DB (Name) H04B 14/06

Claims (1)

(57) [Claim 1] ADPCM data is input and the ADP
Data replacement means for replacing the ADPCM data with a predetermined value so that the absolute value of the CM data is equal to or less than a predetermined threshold value; and a transmission error occurs in the ADPCM data of the frame with the predetermined bit of ADPCM data as one frame. An error signal indicating that the error signal has been input, and threshold setting means for changing and setting the predetermined threshold of the error frame and the subsequent normal frame according to the number of consecutive error frames having the error signal; ADPCM decoder means for decoding an output signal from the data replacement means, an output signal from the ADPCM decoder means, and the error signal are input, and the output signal from the ADPCM decoder means corresponding to the error frame and the subsequent normal frame is input. The level of the signal is determined according to the number of consecutive error signals. Wireless communication device being characterized in that includes the attenuation processing means for performing attenuation processing, the.