JP4401331B2 - Audio processing method and apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice processing means and apparatus capable of reducing deterioration in a received signal under a bad communication environment. <P>SOLUTION: A reception section 10 sequentially receives a plurality of slots, that is, voice data included in second and third slots among last half four slots. Further, a demodulation section 40 respectively demodulates the voice data included in the second and third slots received by the reception section 10 to output voice. When the voice data included in either of the second and third slots are in error, the demodulation section 40 adjusts a volume of the voice in the case of reproducing voice data included in a slot at a post-stage of the slot whose voice data are in error so as to reduce a noise output. <P>COPYRIGHT: (C)2007,JPO&amp;INPIT

Description

  The present invention relates to a signal processing technique, and more particularly to an audio processing method and apparatus for processing a radio signal, and a reception method and apparatus.

  In a mobile communication system, for example, a second generation cordless telephone system, transmission / reception of uplink / downlink signals between a base station and a mobile terminal device is a frame unit composed of a plurality of slots, as defined by the standard. Is done. Conventionally, in each frame, for example, in uplink communication from a plurality of mobile terminal apparatuses to a base station, communication channels from a plurality of mobile terminal apparatuses are assigned one-to-one to a plurality of time-divided slots having different frequencies. Has been done by. However, in each frame, a channel from one mobile terminal apparatus is allocated to only one slot. For example, when the mobile terminal apparatus is far away and the call environment is not good, There is a high possibility that a reception error will occur and a signal from the mobile terminal device cannot be received.

In order to deal with such a problem, conventionally, in each frame, a communication channel from one mobile terminal apparatus is not assigned to one slot as in the prior art, but is assigned to a plurality of, for example, two slots having different timings and frequencies. (See, for example, Patent Document 1). In this way, by transmitting the same information in a plurality of slots having different timings and frequencies, even if the communication environment is bad and a reception error occurs in a certain slot, the same mobile terminal apparatus in any other slot There is a possibility that information of the same content from will arrive at the base station, and the possibility that a signal will be finally received increases.
JP 2002-290299 A

  However, in a poor communication environment, all slots containing the same information are often erroneous, and in such a case, reception performance is deteriorated.

  The present invention has been made in view of such circumstances, and an object of the present invention is to provide an audio processing method and apparatus, and a receiving method and apparatus that can reduce deterioration of a received signal in a poor communication environment.

  In order to solve the above-described problem, an audio processing apparatus according to an aspect of the present invention includes a receiving unit that sequentially receives audio data included in a plurality of slots, and audio data included in the plurality of slots received by the receiving unit. And a demodulator that demodulates and outputs sound. When the audio data included in any of the plurality of slots is incorrect, the demodulator adjusts the volume of the audio when reproducing the audio data included in the slot subsequent to the slot in which the audio data is incorrect.

  According to this aspect, the output of noise included in the reproduced audio can be reduced by adjusting the volume of the subsequent audio data that is affected by the erroneous audio data.

  Each slot of the plurality of slots may include the same audio data. The demodulation unit may include an estimation unit, a first calculation unit, a second calculation unit, a selection unit, and an adjustment unit. The estimation unit may estimate the audio data included in the transmitted plurality of slots for each slot from the audio data included in the plurality of slots received by the reception unit. In addition, the first calculating unit associates the audio data included in the plurality of slots estimated by the estimating unit and the audio data included in the plurality of slots received by the receiving unit with each other while correlating with each other for each slot. The error of the received voice data with respect to the data may be calculated. Further, the second calculation unit may calculate a difference between the errors calculated by the first calculation unit.

  In addition, when the difference calculated by the second calculation unit and the second calculation unit is equal to or greater than a predetermined threshold, the selection unit has a small error among the audio data included in the plurality of slots estimated by the estimation unit. The audio data included in the other slot may be selected and output. In addition, when the difference calculated by the second calculation unit is smaller than the predetermined threshold, the selection unit has a smaller value corresponding to the rear part of the slot among the errors calculated by the first calculation unit. It is also possible to select and output the audio data included in the slot and including the audio data estimated by the estimation unit. In addition, when the error in the rear part of the audio data included in the slot selected by the selection unit is greater than the threshold value relating to the volume adjustment, the adjustment unit is configured to reproduce the audio data included in the slot subsequent to the slot. The sound volume may be adjusted.

  The demodulation unit further includes a detection unit that detects an error for each slot. When errors are detected for all slots in the detection unit, the estimation unit may output the estimation result to the first calculation unit.

  In addition, the demodulator reduces the sound volume when reproducing the sound data included in the slot subsequent to the slot in which the sound data is incorrect when the sound data included in any of the plurality of slots is incorrect. May be.

  According to this aspect, even if all the audio data is incorrect, the slot including the audio data with few errors is selected and output, so that the noise output can be reduced.

  Another aspect of the present invention is a receiving device. The apparatus includes a receiving unit, an estimating unit, a first calculating unit, a second calculating unit, and a selecting unit. The receiving unit receives a plurality of transmitted same data. In addition, the estimation unit estimates each of the plurality of transmitted data from the plurality of data received by the reception unit. The first calculation unit calculates an error of the received data with respect to the estimated data while associating the estimation unit, the plurality of data estimated by the estimation unit, and the plurality of data received by the reception unit. Further, the second calculation unit calculates a difference between the errors calculated by the first calculation unit and the first calculation unit. In addition, when the difference calculated by the second calculation unit is equal to or greater than a predetermined threshold, the selection unit selects and outputs data having a smaller error among the plurality of data estimated by the estimation unit. In addition, when the difference calculated by the second calculation unit is smaller than the predetermined threshold, the selection unit has a smaller value corresponding to the rear part of the data among the errors calculated by the first calculation unit. Select and output data that is data and estimated by the estimation unit.

  According to this aspect, even if all the data are erroneous, it is possible to select and output data with few errors. Therefore, stable reception performance can be obtained.

  Yet another embodiment of the present invention is a voice processing method. This method includes the steps of sequentially receiving audio data included in a plurality of slots, and demodulating the audio data included in the plurality of slots received in the receiving step and outputting the audio. In the step of outputting sound, when the sound data included in any of the plurality of slots is incorrect, the sound volume when reproducing the sound data included in the slot subsequent to the slot where the sound data is incorrect is set. Adjust.

  According to this aspect, the output of noise included in the reproduced audio can be reduced by adjusting the volume of the subsequent audio data that is affected by the erroneous audio data.

  Yet another embodiment of the present invention is a reception method. The method includes receiving a plurality of transmitted identical data, estimating a plurality of transmitted data from a plurality of data received in the receiving step, and estimating a plurality of estimated data in the estimating step. While associating the data with a plurality of data received at the receiving unit, the difference between the errors calculated in the first step of calculating the error of the received data with respect to the estimated data and the first step of calculating is calculated. If the difference calculated by the second step and the second step calculation unit to be calculated is equal to or greater than a predetermined threshold value, the data with the smaller error is selected and output from the plurality of data estimated by the estimation unit If the difference calculated in the second step of calculation is smaller than the predetermined threshold value, it is calculated in the first step of calculation. Of errors in les, comprising the steps of a data whichever value of a portion corresponding to the rear portion of the data is smaller, and selects and outputs the data estimated by the estimating unit.

  According to this aspect, even if all the data are erroneous, since data with few errors is selected and output, noise output can be reduced.

  Another aspect of the present invention is an audio processing method. The method includes a receiving step, a detecting step, an estimating step, a first calculating step, a second calculating step, a selecting step, and an adjusting step. The receiving step sequentially receives the same audio data included in each of the plurality of slots. The detecting step sequentially detects errors included in the audio data. In the step of estimating, when errors are included in all the audio data in the detecting step, the audio data included in the plurality of slots transmitted from the audio data included in the plurality of slots is respectively determined for each slot. presume. In the first calculation step, the estimated voice data included in each slot is associated with the voice data included in the plurality of slots estimated in the estimating step and the voice data included in the plurality of slots received in the receiving step. The error of the received voice data with respect to the data is calculated. In the second calculation step, the difference between the errors calculated in the first calculation step is calculated. In the selection step, when the difference calculated in the second calculation step is greater than or equal to a predetermined threshold value, the audio included in the slot with the smaller error among the audio data included in the plurality of slots estimated in the estimating step Select and output data. In the selection step, when the difference calculated in the second calculation step is smaller than a predetermined threshold value, the value corresponding to the rear portion of the slot is smaller in each error calculated in the first calculation step. The audio data included in the slot and including the audio data estimated in the estimating step is selected and output. In the adjustment step, when the error in the rear part of the audio data included in the slot selected in the selection step is larger than the threshold value related to the volume adjustment, the audio data for reproducing the audio data included in the subsequent slot of the slot Adjust the volume.

  According to this aspect, even if all data is wrong, by adjusting the volume of the subsequent audio data that is affected by the erroneous audio data, output of noise included in the reproduced audio Can be reduced.

  It should be noted that any combination of the above-described constituent elements and a conversion of the expression of the present invention between a method, an apparatus, a system, a recording medium, a computer program, etc. are also effective as an aspect of the present invention.

  According to the present invention, it is possible to reduce deterioration of a received signal even in a poor communication environment.

  Before describing the present invention specifically, an outline will be given first. An embodiment of the present invention relates to reception and demodulation of voice data in a terminal device of a second generation cordless telephone system. The terminal device includes a receiving unit that receives audio data and a demodulating unit that performs demodulation processing on the received audio data. Here, it is assumed that the received audio data includes a plurality of identical audio data and is transmitted over a plurality of slots. If the audio data in all the slots is incorrect, the slot with the least error is selected and output. The small number of errors is derived by calculating the magnitude of the error of the audio data included in the rear part of the slot among the audio data included in the slot. In addition, even in the slot with fewer errors, there is still an error, so the magnitude of the error is compared with the threshold for volume adjustment, and if there is an error exceeding the threshold, The volume of the audio data contained in the slot is adjusted for a slot subsequent to the slot containing the audio data having an error.

  Here, the reason for adjusting the volume of the audio data contained in the slot subsequent to the slot containing the erroneous audio data is as follows. That is, when the audio data is encoded using differential encoding such as adaptive differential pulse code modulation (hereinafter abbreviated as “ADPCM”), This is because if the slot is incorrect, the error is affected by the subsequent slot.

  Here, ADPCM is a compression method of voice data that is often used for voice coding of mobile phones and second generation cordless telephone systems. While pulse code modulation (hereinafter abbreviated as “PCM”) is simply recorded by sampling and quantization, ADPCM takes a difference from the immediately preceding data, Furthermore, compression is performed by performing adaptive quantization. Since the voice changes continuously, if the difference is taken, it becomes a smaller value than the original value, and therefore it can be expressed with a smaller amount of data. In addition, by performing adaptive quantization, it is possible to record not only a large change but also a small change.

  Since ADPCM is a method of compressing the audio data by taking the difference from the immediately preceding audio data as described above, if the immediately preceding audio data has an error, the audio data is also erroneous. However, as in this embodiment, when there is an error in the previous stage data, the output of noise can be suppressed by lowering the volume of the subsequent stage data. If there is no error in the preceding data, the volume of the succeeding data may be increased.

  Hereinafter, the configuration and operation of the present embodiment will be described in detail with reference to the drawings.

  FIG. 1 is a diagram illustrating a configuration example of a terminal device 100 according to an embodiment of the present invention. The terminal device 100 of FIG. 1 may be, for example, a mobile phone or a terminal used in a second generation cordless telephone system. The terminal device 100 includes a reception unit 10 that receives transmitted audio data, a demodulation unit 40 that demodulates audio data received by the reception unit 10, a reproduction unit 50 that reproduces audio demodulated by the demodulation unit 40, and a reception A reproducing unit 50 for controlling the unit 10, the demodulating unit 40, and the reproducing unit 50. Here, the audio data received by the receiving apparatus of FIG. 1 will be described with reference to FIG.

  FIG. 2 is a diagram illustrating a frame configuration example of a signal received by the terminal device 100 of FIG. In the frame configuration example of FIG. 2, the horizontal axis is time, and in the frame configuration consisting of 8 slots, the first 4 slots are slots for performing upstream signal transmission processing, and the last 4 slots are downstream signals. This is a slot for performing reception processing. In the first four slots and the latter four slots, control signals (Control Channels, hereinafter abbreviated as “CCH”) are transmitted / received in the respective first slots. In each of the second slot and the third slot, voice data of a certain user is transmitted / received. Here, in the second slot and the third slot, the users to be transmitted / received are the same, and the audio data to be transmitted / received are the same. Also, in each fourth slot, the same audio data for the same user may be the same as in the second slot and the third slot, and the target users in the second slot and the third slot are Different users may be sent / received. Note that the terminal device that is the object of the present embodiment receives and processes the downstream signals of the latter four slots in FIG.

  Here, the receiving unit 10 of FIG. 1 sequentially receives audio data included in a plurality of slots, that is, the second slot and the third slot in the latter half 4 slots. Next, the demodulator 40 demodulates the audio data contained in the second slot and the third slot in the last four slots received by the receiver 10 and outputs audio. The demodulator 40 selects and outputs audio data with no error when the audio data contained in either the second slot or the third slot in the latter half 4 slots is incorrect. When all the audio data is incorrect, the volume of the audio when reproducing the audio data included in the slot subsequent to the slot in which the audio data is incorrect is adjusted to reduce the noise output. Further, if the audio data included in either the second slot or the third slot is not in error, the sound volume when reproducing the audio data included in the slot subsequent to the slot may be increased. Good. In the following, in order to simplify the description, “the second slot (and the third slot) in the latter four slots” is simply abbreviated as “the second slot” or “the third slot”. To do.

  The reproduction unit 50 reproduces the sound demodulated by the demodulation unit 40. Specifically, the audio decoding process corresponding to the audio encoding process performed on the transmission side is performed, and the audio is reproduced via a speaker or the like. At this time, the volume may be adjusted and output.

  The control unit 60 controls the receiving unit 10, the demodulating unit 40, and the reproducing unit 50. Specifically, the control unit 60 exchanges control signals such as reception timing, control parameters, and volume adjustment amounts among the reception unit 10, the demodulation unit 40, and the reproduction unit 50.

  Here, the configuration of the demodulator 40 will be described in detail with reference to FIG. FIG. 3 is a diagram illustrating a configuration example of the demodulation unit 40 in FIG. The demodulator 40 includes an estimator 12, a detector 14, a first calculator 20, a second calculator 22, a selector 30, an adjuster 32, and an output unit 34.

  The estimation unit 12 estimates, for each slot, the audio data included in the transmitted plurality of slots from the audio data included in the plurality of slots received by the reception unit 10. Specifically, demodulation processing corresponding to the modulation processing performed on the transmission side is performed on the audio data received by the receiving unit 10. Further, deinterleaving processing corresponding to the interleaving processing performed on the transmission side may be performed. Furthermore, error correction decoding corresponding to the error correction coding performed on the transmission side may be performed. Both processes are performed for each slot. When the signal having the frame configuration shown in FIG. 2 is received, the above-described processing is performed for each of the second slot and the third slot. The estimated result may be output as a hard decision value or as a soft decision value.

  The detection unit 14 detects an error as to whether or not the audio data for each slot estimated by the estimation unit 12 includes an error. Error detection may be performed by using error detection control data, for example, cyclic redundancy check codes, attached to audio data or inserted between audio data sequences. . In the error detection, when errors are detected for all slots, the result estimated by the estimation unit 12 is output to the first calculation unit 20. If there is even one slot with no error, the estimated voice data included in the slot without the error is output to the output unit 34. When a signal having the frame configuration illustrated in FIG. 2 is received, if an error is included in the estimation results for both the second slot and the third slot, both estimation results are used as the first calculation unit 20. Output to. If there is no error in either the second slot or the third slot, the slot without error is output to the output unit 34. The other slot may be discarded.

The first calculation unit 20 estimates the audio data included in the plurality of slots estimated by the estimation unit 12 and the audio data included in the plurality of slots received by the reception unit 10 while associating each of the slots with each other. An error of the received voice data with respect to the voice data is calculated. A case where a signal having the frame configuration shown in FIG. 2 is received will be described as an example. Here, the audio data estimated for the second slot in the estimation unit 12 to soft decision value of the obtained sequence and A _2. Also, it was soft decision value of the speech data estimated for the third slot in the estimation unit 12 series and A _3. Further, the sequence of voice data received by the receiver 10 and B _2. A series of audio data received by the receiver 10 and B _3. Here, the soft decision value conversion means converting the hard / soft decision value X into the soft decision value Y according to the equation (1).
Y = (1-2X) × (half value of dynamic range of AD converter) Expression (1)

  Here, “the half value of the width of the dynamic range of the AD converter” is, for example, a case where the AD converter is composed of 6 bits, and if the dynamic range is −32 to +31, the width is 63, and 31.5 of the half value becomes “a half value of the width of the dynamic range of the AD converter”.

Further, if the sequences A ₂ , A ₃ , B ₂ , and B ₃ are sequences having N symbols, they are expressed as follows.
A ₂ = {a ₂₀ , a ₂₁ ,..., A _{2 (N−1)} } Expression (2)
A ₃ = {a ₃₀ , a ₃₁ ,..., A _{3 (N−1)} } Expression (3)
B ₂ = {b ₂₀ , b ₂₁ ,..., B _{2 (N−1)} } Expression (4)
B ₃ = {b ₃₀ , b ₃₁ ,..., B _{3 (N−1)} } Expression (5)

  Here, in order to simplify the explanation, the coding rate in error correction coding is assumed to be 1, and the lengths of all the sequences are assumed to be equal. Then, the error E of the received voice data with respect to the estimated voice data can be expressed as follows.

^{_{e 2k = SQRT ((B 2kI}} -A 2kI) 2 + (B 2kQ -A 2kQ) 2)) ··· formula (8)
e _3k = SQRT ((B _3kI −A _3kI ) ² + (B _3kQ −A _3kQ ) ² )) (9)

Here, B _2kI , A _2kI , B _3kI , and A _3kI are k-th I signals in the respective slots. B _2kQ , A _2kQ , B _3kQ , and A _3kQ are k-th Q signals in the respective slots. SQRT is an operator for calculating the square root of the numerical value in parentheses.

The second calculation unit 22 calculates the difference between the errors calculated by the first calculation unit 20. In the above example, when the difference is ΔE, the calculation is as follows.
ΔE = | E ₃ −E ₂ | Expression (10)

  When the difference calculated by the second calculation unit 22 is equal to or greater than the predetermined threshold T, the selection unit 30 selects the slot with the smaller error among the audio data included in the plurality of slots estimated by the estimation unit 12. Select and output included audio data. On the other hand, when the difference calculated by the second calculation unit 22 is smaller than the predetermined threshold value, among the errors calculated by the first calculation unit 20, the value corresponding to the rear part of the slot is the smaller slot. A slot that is included in the audio data and includes the audio data estimated by the estimation unit 12 is selected and output.

Specifically, when the value of ΔE is equal to or greater than T when the threshold value T and the difference ΔE are compared, the audio data included in the slot corresponding to the smaller error of E ₃ and E ₂ is Select and output. For example, towards the E ₃ it may E ₂ less, so that the third slot of the audio data is output.

On the other hand, when the value of ΔE is smaller than the threshold value T, “the value of the portion corresponding to the rear portion of the slot” is obtained. The “rear part of the slot” may be the last part when the slot is divided into three parts, or may be the latter half part when the slot is divided into two parts. Here, a case where the slot is divided into three will be described. Here, when E ₂ and E ₃ are divided into three, formulas (6) and (7) are rewritten as follows.
_{E 2 = E 2PRE + E 2MID} + E 2POST ··· formula (11)
_{E 3 = E 3PRE + E 3MID} + E 3POST ··· formula (12)

Here, E _2PRE and E _3PRE are the errors in the forefront when each slot is divided into three. E _2MID and E _3MID are errors of the second part from the front when each slot is divided into three. E 2 _POST and E 3 _POST are errors of the third part from the front when each slot is divided into three. Here, assuming that N = 3 × M (M is a positive integer), E _2PRE , E _2MID , E _2POST , E _3PRE , E _3MID , and E _3POST are expressed as follows.

To determine whether there is an error in the rear of each slot, for the second _slot, and _{E 2POST, E 2PRE,} or _checks the magnitude relationship between _{E 2MID,} when _{E 2POST} largest, 2 It is determined that there is an error at the rear of the slot. Similarly, the third slot _determines the _{E 3POST, E 3PRE} or _checks the magnitude relationship between _{E 3MID,} when _{E 3POST} largest, to the rear of the third slot and there is an error. In addition, in order to determine whether or not there is an error in the rear part of each slot, it may be obtained by comparing with a predetermined threshold value using E _2POST and E _3POSt .

  Finally, the selector 30 compares the error in the rear part of the second slot with the error in the rear part of the third slot, and selects and outputs the audio data contained in the slot with the smaller error.

  When the error in the rear part of the audio data included in the slot selected by the selection unit 30 is larger than the threshold value related to the volume adjustment, the adjusting unit 32 reproduces the audio data included in the subsequent slot of the slot. Reduce the volume of the audio. If the error at the rear of the audio data is smaller than the threshold value for volume adjustment, the volume may be increased. Here, the “rear error” is determined as described above, and the volume is adjusted as compared with a threshold relating to volume adjustment.

  Here, the operation of the receiving apparatus of FIG. 1 will be described in detail with reference to FIG. FIG. 4 is a diagram illustrating an operation example of the receiving apparatus of FIG.

  First, the receiving unit 10 sequentially receives audio data included in a plurality of slots (S10). Next, the estimation unit 12 estimates the audio data included in the transmitted plurality of slots for each slot from the audio data included in the plurality of slots received by the reception unit 10 (S12).

  The detection unit 14 detects an error for each slot in the results estimated by the estimation unit 12 (S14). If errors are detected for all slots as a result of detection by the detection unit 14 (Y in S16), the result estimated by the estimation unit 12 is output to the first calculation unit 20. On the other hand, if no error is detected in any one slot (N in S16), the audio data included in the slot in which the error is not detected is selected and output (S18). If no error is detected for two or more slots, any one of the slots may be selected and audio data included in the slot may be output.

  Next, the first calculation unit 20 associates the audio data included in the plurality of slots estimated by the estimation unit 12 with the audio data included in the plurality of slots received by the reception unit 10 for each slot. Then, the error of the received voice data with respect to the estimated voice data is calculated (S20). The error is calculated by the method described above. For example, when a signal having a frame configuration as illustrated in FIG. 2 is received, the calculation may be performed using Expressions (1) to (9).

  Next, the second calculation unit 22 calculates the difference between the errors calculated by the first calculation unit 20 (S22). When a signal having a frame configuration as illustrated in FIG. 2 is received, the calculation may be performed using Equation (10). When there are slots including audio data of three slots or more, the difference between the errors is calculated. In such a case, since there are three combinations for selecting two of the three, three differences are calculated.

Next, each difference calculated by the second calculation unit 22 is compared with a predetermined threshold value. If any of the differences is larger than the predetermined threshold (N in S24), the audio data included in the slot having the smallest error among the errors calculated by the first calculation unit 20 in S20 is selected. And output (S26). When a signal having a frame configuration as shown in FIG. 2 is received and ΔE obtained by Equation (10) is larger than a predetermined threshold value, the smaller one of E ₂ and E ₃ Is selected. Here, E ₂ if E ₃ is smaller than, and the second slot of the audio data is selected.

  On the other hand, when all the differences are smaller than the predetermined threshold (Y in S24), the selection unit 30 has a small value corresponding to the rear portion of the slot among the respective errors calculated by the first calculation unit 20. The slot containing the voice data included in the other slot and including the voice data estimated by the estimation unit 12 is selected and output (S30). When a signal having a frame configuration as shown in FIG. 2 is received, the rear error may be calculated using equations (13) to (18).

  Next, when the error in the rear part of the audio data included in the slot selected by the selection unit 30 is larger than the threshold value relating to the volume adjustment, the adjustment unit 32 reproduces the audio data included in the slot subsequent to the slot. The volume of the sound at the time of performing is reduced (S32). If the threshold is smaller than the threshold for adjusting the volume, the volume may be increased.

  According to the embodiment of the present invention, it is possible to reduce deterioration of a received signal in a poor communication environment. Further, by adjusting the volume of the subsequent audio data that is affected by the erroneous audio data, the output of noise included in the reproduced audio can be reduced. By reducing the output of noise, the output of abnormal noise or the like can be reduced, and a comfortable use environment can be provided to the user. Further, by selecting audio data included in a slot with few errors from audio data, it is possible to reproduce good audio. Also, better audio can be reproduced by determining errors by focusing on the latter half of the audio data.

  In the above, this invention was demonstrated based on the Example. This embodiment is an exemplification, and it will be understood by those skilled in the art that various modifications can be made to the combination of each component and each processing process, and such modifications are also within the scope of the present invention. .

It is a figure which shows the structural example of the terminal device concerning the Example of this invention. It is a figure which shows the example of a frame structure of the signal which the terminal device of FIG. 1 receives. It is a figure which shows the structural example of the demodulation part of FIG. It is a figure which shows the operation example of the terminal device of FIG.

Explanation of symbols

  10 receiving units, 12 estimating units, 14 detecting units, 20 first calculating units, 22 second calculating units, 30 selecting units, 32 adjusting units, 34 output units, 40 demodulating units, 50 reproducing units, 60 control units.

Claims (3)

A receiving unit for sequentially receiving the same audio data included in each of the plurality of slots;
A demodulator that demodulates audio data contained in a plurality of slots received by the receiver and outputs audio;
The demodulator
An estimation unit that estimates, for each slot, audio data included in a plurality of transmitted slots from audio data included in a plurality of slots received by the reception unit;
The audio data received for the estimated audio data while the audio data included in the plurality of slots estimated by the estimation unit and the audio data included in the plurality of slots received by the reception unit are associated with each other for each slot. A first calculation unit for calculating respective errors of
A second calculation unit for calculating a difference between the errors calculated by the first calculation unit;
If the difference calculated by the second calculation unit is greater than or equal to a predetermined threshold, the audio data included in the slot with the smaller error is selected from the audio data included in the plurality of slots estimated by the estimation unit When the difference calculated by the second calculation unit is smaller than a predetermined threshold value, the value corresponding to the rear part of the slot is smaller in each error calculated by the first calculation unit. A selection unit that selects and outputs a slot that includes the voice data included in the slot and includes the voice data estimated by the estimation unit;
When the error in the rear part of the audio data included in the slot selected by the selection unit is larger than the threshold for adjusting the sound volume, the sound volume when reproducing the audio data included in the slot subsequent to the slot is adjusted. An audio processing apparatus comprising: an adjustment unit that performs the adjustment. The demodulator
A detector for detecting an error for each slot;
2. The speech processing apparatus according to claim 1, wherein when an error is detected for all slots in the detection unit, the estimation unit outputs an estimation result to the first calculation unit. Sequentially receiving the same audio data respectively contained in a plurality of slots;
Sequentially detecting errors included in the audio data;
When all the audio data includes an error in the detecting step, the audio data included in the plurality of slots is estimated for each slot from the audio data included in the plurality of slots, respectively. ,
The audio data included in the plurality of slots estimated in the estimating step and the audio data included in the plurality of slots received in the receiving step are associated with each other for each slot, and the estimated audio data is received. A first calculation step for calculating each error of the audio data;
A second calculation step of calculating a difference between the errors calculated in the first calculation step;
If the difference calculated in the second calculating step is greater than or equal to a predetermined threshold value, the audio data included in the slot with the smaller error among the audio data included in the plurality of slots estimated in the estimating step When the difference calculated and output in the second calculation step is smaller than a predetermined threshold value, the value of the portion corresponding to the rear portion of the slot is small among the respective errors calculated in the first calculation step. A selection step for selecting and outputting a slot that is voice data included in the other slot and includes the voice data estimated in the estimation step;
If the error in the rear part of the audio data included in the slot selected in the selection step is larger than the threshold for adjusting the sound volume, the sound volume when reproducing the audio data included in the slot subsequent to the slot is adjusted. An audio processing method comprising: an adjusting step.

Images