TW200410508A - Method for detecting a tone signal through digital signal processing - Google Patents

Abstract

A method for detecting a tone signal having a predetermined tone frequency includes receiving an analog communication signal, using a sampling frequency to convert the analog communication signal into a digital communication signal, selecting a plurality of signal sectors, which corresponds to a predetermined frequency range, out of the digital communication signal wherein the predetermined tone frequency agrees with the predetermined frequency range, and using the number of samples of each signal sector to determine whether the analog communication signal contains the tone signal.

Description

200410508 V. Description of the Invention (l) Technical Field of the Invention The present invention provides a method for a wireless communication device to detect an audio signal corresponding to a tone frequency, especially a wireless communication device using digital signal processing (digital signal). processing) and a zero-crossing rate (Zero-crossing rate, ZCR) process to extract and measure the audio signal. The prior art continuous tone-coded squelch system (CTCSS) has been widely used in the field of wireless communication, and is mainly used to allow multiple users to pass through the same communication channel (cha η) in a communication area. ne 1) For a call, the continuous audio coding mute system uses a low-frequency audio signal (CTCSS tone) to distinguish it from a signal transmitted on a communication channel. For example, walki-talki Continuous audio coding mute system to achieve the purpose of group communication. Please refer to Figure 1. Figure 1 is a schematic diagram of the frequency range used in the conventional continuous audio coding mute system. As shown in Figure 1, the frequency range between frequencies 62 · 5Hz and 250Hz is mainly used to transmit the above-mentioned low-frequency audio signals, and the frequency range between frequencies 〇〇Ηz and 3.4KΗζ is used to transmit A user generates a speech signal. For the conventional portable wireless walkie-talkie, the operation principle is briefly described as follows. Generally, 14 communication channels are used.

200410508

V. Description of the invention (2) (channel) Pf " · The Pi bed transmits and receives signals. The communication channel is a physical channel. In addition, each communication channel can be based on 38 different frequencies. The audio signal T f ••… T 3 涞 generates a total of 532 (14 * 38) logical channels. When a speaker sets the physical channel used by the portable radio as P slave and the audio signal used Is T !, that is, the caller sets its logical channel to P i (T 〇, when the caller presses the push-to-talk (PTT) button of the portable radio, the caller can pass Read the physical channel P to output its voice signal to the predetermined communication range corresponding to the portable radio, if there are three other users in the predetermined communication range, and set the logic used by them

^ ^ ^ w ^ ^ 1 1 V 1 1; 'ϊ ^ 1 38; Λ (το, for the first user, because his portable wireless intercom is connected to the physical channel P for signal transmission and reception, so The first portable wireless intercom used to start receiving the voice output by the speaker ΪαΪ The d intercom judged that the audio signal used by the speaker was also T r, that is, the child speaker and the first one used Channel P 丨 (τ 丨), so if I use the I spoon to recognize the same logic, I will receive the voice message of the caller from -1 to 4 from your Tianben, taxi to the sender Voice signal for the second user, because a portable walkie-talkie is attached to the second user

Line signal transmission and reception, so the second user using hand m will also start to receive the voice signal output by the caller. Ίί 4 line radio to determine the audio signal used by the caller? The two users were numbered as $ 泥 冯 11 instead of τ38. Different logical channels

Page 6 200410508 V. Description of the invention (3) (Ti) and Pι (Τ38), so the machine will not receive it. The second person uses / hand-held wireless intercom so the second user if The voice = number i is output by one t eight, and the voice signal of the third user and t, d? Is transmitted and connected to the final signal of the channel P. The wireless f walkie-talkie is connected to the physical walkie-talkie. Use the same wireless connection with the speaker as 甬 to make the second wireless mobile user's portable wireless pair 'hence the third / · "· but the two machines cannot follow the C received signal st 丄! · "· The eighth indicator (RSSI) detected f, the signal strength to start receiving the signal sent by the caller, so the portable radio of the third user will not receive the voice signal of the caller, Two is also = the speaker's voice signal will be output through a speaker, so the second and third users cannot hear the voice signal of the caller, that is, the portable radio of the second and third users passes Judging the audio signal and activating the Squelch function, in other words, users with the same logical channel can talk to each other to achieve the purpose of the above group communication. Therefore, for the receiver, how to test the debt The logical channel used by the speaker has become an important subject for conventional portable radios.

Please refer to FIG. 1 and FIG. 2 'FIG. 2 is a functional block diagram of a conventional portable radio 10. The portable wireless walkie-talkie 10 includes an antenna U, a transceiver 12, a selector 14, a CTCSS decoder 16, a CTCSS encoder 18, and a voice signal. Processing unit 20, 〆

Page 7 63Q 200410508 V. Description of the Invention (4) System ^ fDP, aker) 22 ’A microphone (Aerophone) 24, and the value of 2 U Danfan 26. The portable wireless walkie-talkie 10 can receive and transmit the secondary frequency signal RF via the antenna 11. For receiving the radio frequency signal RF, the transceiver 12 converts the = wh radio frequency signal RF into a low-frequency fundamental frequency signal Rx, and transmits it to Select the I state to, and then the selector 14 will output the baseband signal x through the output terminal A, and the decoder 16 will judge the frequency of its audio signal based on the baseband signal ". Generally speaking, the decoder 1 6 contains an analog filter circuit to pass the signal of the base frequency signal Rx between the frequency of 62.5 Hz and the frequency of 250.112 to the audio signal corresponding to the base frequency signal ^ to determine whether to use phase =, The decoder unit 16 will judge it 26 at the same time. If the baseband signal is set to the same logical channel as the corresponding channel, the control unit will use: to enter the light after the # signal output processing, the voice signal processing unit There are also two analog filter circuits, which are used to capture the signal between the high-pass crossing waves with a frequency of 30 0H3 and a rate of 3.4KΗζ, and then pass through ^ ff. Conversely, if the fundamental frequency signal Rx corresponds to the logical channel Toilet = 2 sets of logic channels set different control rules 2, 6 will not; ^ $, the voice signal processing unit 20 and the speaker 22, so the hand ^ will not output any transmitted through different logical channels; J pair = 〇, radio frequency signal RF ... When-the user presses the 14 will select the input terminal B 'and the control unit 26 will start / $ opt / $ The user's $ audio signal can be input to the voice signal. As mentioned earlier, the voice signal processing unit 20 filters through its filter. Μ, the signal between the rate of 3 0 0 I1Z and the frequency of 3 · 4Κ ， ζ, after the bran • the circuit to get the frequency ..., and then output to the encoder

200410508 V. Description of the invention (5) 18, the encoder 18 will generate the basic signal by adding the corresponding audio signal to the output signal of the voice signal processing unit 20 according to the logical channel (CTCSS code) set by the portable radio. The frequency signal τχ is finally converted into a high-frequency radio frequency signal RF by the transceiver 12 and output through the antenna 11. As mentioned above, the conventional portable radios are performing analog signal processing, that is, for receiving analog radio frequency signals RF to outputting analog baseband signals Rx via π 22 22, the analog = formula is used. For audio signals and other related processing, that is, the decoder 6 must use a conventional filter circuit to capture the required frequency range (6.2 to 2.5 Hz). However, the conventional filter circuit cannot be obtained due to its characteristics. Accurate (sharp) filtering characteristics to extract signals in the required frequency range. For example, when 38 audio signals are evenly distributed between the frequency 62 · 5Hz to 25 0KHz, the frequency difference between adjacent audio signals is only about 3 To 5Hz, that is, the decoder 16 will most likely produce misjudgment audio signals and affect the actual signal reception. SUMMARY OF THE INVENTION Therefore, a main object of the present invention is to provide a method for a wireless communication device to detect an audio signal corresponding to an audio frequency by using digital signal processing and zero-crossing rate processing to solve the above problems. The scope of this patent application provides a method for a wireless communication device to detect an audio signal corresponding to a tone frequency.

Page 9 200410508 V. Description of the invention (6) It includes receiving an analog voice signal and using a sampling frequency to convert the analog voice signal into a digital w saying that the digital $ 我 音 机 号 系It is composed of a plurality of signal segments, and each signal segment corresponds to a period (period); within a frame time, a plurality of first digital speech signals corresponding to a predetermined frequency range are selected. A signal segment; and determining whether the analog voice signal includes an audio signal corresponding to an audio frequency according to the number of sample points corresponding to each first signal segment. The audio frequency is located in the predetermined frequency range. Embodiment Please refer to FIG. 3, which is a schematic diagram of a functional block of the decoder 30 of the present invention. The decoder 30 is applied to the conventional portable wireless walkie-talkie 10 shown in FIG. 2 to detect the corresponding audio signal of the communication signal output by a speaker to determine whether to use the same logical channel. In this embodiment, the decoder 30 includes an analogue-to-digi tal converter (ADC) 32, a digital signal processor (DSP) 34, and a random access memory. Random access memory (RAM) 36, and a read-only memory (ROM) 38. As shown in FIG. 2, when the conventional portable radio 1 10 receives a voice signal A1 via the antenna 11 and transmits it to the decoder 30 for audio signal processing, the analog / digital converter 3 2 of this embodiment is then processed. Will take the analog voice signal A1 at a sampling frequency

Page 10 200410508 V. Description of the invention (7) (samp 1 ing frequency) is converted to a digital voice signal m, and the voice signal D 1 is further detected by the digital signal processor 3 4 to determine whether the voice signal D 1 is Use the logical channel set by the portable wireless walkie-talkie 10, that is, the digital signal processor 3 4 series processes the voice signal d 1 corresponding to the frequency 6 2 · 5 Η z ~ 2 5 0 Η z to determine the voice signal a 1 corresponds to the tone frequency. In addition, the read-only memory 38 stores the calculation program required by the digital signal processor 34 to perform the above operations, and temporarily stores the digital signal processor 40 through the random access memory 36 to execute the calculation program. The generated temporary data, that is, the random access memory 48 is a data buffer. When the digital signal processor 34 finishes processing the audio signal of the voice signal D1, a detection result will be generated! ) 2, and transmitted to a control circuit 40 (such as a microprocessor), which is used to control the overall operation of the conventional portable wireless trencher and provides a man-machine interface (MMI) to enable A user learns the state of the portable radio through the human-machine interface and operates the portable radio through the human-machine interface, so the control circuit 40 judges whether the portable radio has passed a The output device (for example, 8 2 2 left in Fig. 2) outputs a signal with a frequency between 300 Hz and 3.4 kHz in the voice signal A 丨 to complete signal reception. Please refer to FIG. 4, which is a flowchart of the operation of the digital signal processor 34 shown in FIG. The operation of the digital signal processor 34 includes the following steps: Step 10: Start;

Page 11 634 200410508 V. Description of the invention (8) Step 1 0 2: Perform a pair of logical channel band-pass filtering on one sampling point of the voice signal D 1; Step 1 0 4 ·· A smooth filtering process is performed on the band-passed signal; step 106: use a zero crossing rate (ZCR) process to determine whether the voice signal D1 has been input for a period The sampling point to the digital signal processor 34, if yes, then go to step 10; otherwise, go to step 11 2; step 108: detect the signal energy corresponding to the sampling point in the period, and compare The signal energy and a predetermined energy value are used to determine whether the sampling points in the period are valid (va 1 id), and the number of sampling points in the period is compared with a predetermined count value to determine whether the sampling points in the period are valid. If so, then Go to step 1 1 0, otherwise go to step 112; step 1 1 0: record the number of sampling points in the period; step 112 check whether the frame period has been reached, and if so, execute Step 11 14, otherwise execute step i 〇 2; Step 1 14: Calculate the average value of the number of sampling points in each cycle of the brother record; Step 1 16: Calculate the offset between the average value and the predetermined count value; Step 118: Compare the offset with a threshold value to determine whether the voice signal D1 uses the logical channel, and inform the control circuit, and then return to step 102. The operation of the digital signal processor 3 4 to determine the audio signal is described below, such as

$ 12 页 200410508 V. Description of the invention (9) As shown in Figure 1, the frequency of the audio signal is between 62.5Hz ~ 2500Hz, so the audio signals of 38 different frequencies will be distributed at 62.5Hz ~ 2 5 0 Hz, so this embodiment first uses 38 frequency ranges to filter the input voice signal D1 to simplify the subsequent processing complexity (complexity). For example, if a user sets the logic of his portable radio, The channel is Pι (τ38) ', that is, the 38th audio signal (with a frequency of 25 Ηζ) is used to distinguish it from the various voice signals transmitted by the physical channel P. Therefore, this embodiment will start to respond to the 38th audio signal. The frequency range is 238Ηζ ~ 25〇Ηζ to filter, and the rate range is 2 38Hz ~ 25 0Hz for voice signals outside to improve the processing efficiency of subsequent audio signal detection, that is, the digital signal processor 34 will perform the input voice signal D1. Band-pass filtering (step > 10.2) processing to extract signals that may be related to the audio signal (whose frequency is 25 Hz). Therefore, in this embodiment, 38 frequency ranges are set so as to filter out 38 corresponding audio signals Ti ~ T38 respectively. In other words, in this implementation, one of the corresponding frequencies is activated according to the logical channel set by the user. The two ranges are arranged in a row. Signals that are not related to the corresponding audio signal of the logical channel. Because the signal transmission process between the receiver and the receiver may be affected by ring structures, weather, etc.), it may cause noise interference. =, The waveform of the voice signal A1 received by the interphone will be changed due to the irregular waveform such as the noise η vs. the dog wave (SPlke), so this implementation of the "up = i / Sm〇〇thing) processing (step 104)来 消 = Γ ', Er Er said δΤΙ said the impact of A1. Please refer to FIG. 5. FIG. 5 is a diagram of Γ = 2 1: the waveforms of the audio signal A1 and the digital voice signal D1 ′.枳 car represents time, and the vertical axis represents the sampled value. The voice signal ai is

200410508 V. Description of Invention (10)

The analog signal is obtained by sampling the speech signal A1 at a sampling frequency through the analog / digital converter 32 to generate a speech signal D1. The speech signal D1 is a digital signal, which corresponds to a plurality of sampling points (samPle), as shown in Figure 5. As shown, the speech signal D1 includes a plurality of signal segments 3, S2, and S3, and each signal segment S1, S2, and S3 corresponds to a period (Peri 0 d). However, in the signal segment S2, Between the sampling point SPn— and the sampling point SPn + 由于, because the speech signal A1 is disturbed and generates a surge, the waveform is irregularly changed. Therefore, in theory, the sampling value of the sampling point SP should be a positive value. The effect of the wave becomes a negative value, that is, the sampling point SP 妁 sampling value generates an error. Therefore, the present invention uses a plurality of sampling points before each sampling point to perform a continuous calculation of the average sampling value to make each sampling point confusing. For example, for the sampling point SP n affected by noise, the average value of the sampling points SP n-4, SP n_3, SP n_2, and SP nP SP A can be calculated to update the sampling. Point SP 妁 sample value, so take Sampling point value SP peg in the original scrambled by Rate, so when a positive value indicates a sample point sampling points and sample values

Correcting the wrong negative value to the correct positive value first to reduce the noise. In this embodiment, the start and end of each signal section is judged in the manner of the known zero-crossing ZCR. When the sampling values of adjacent sampling points change between positive and negative values, that is, the end of the half cycle and the beginning of the other half cycle, as shown in Figure 5, the sampling value of s ρ is negative, and the next sampling point sp趵 The sample value is 'so the sample point Spm is the last one in the signal section S2 and the sample point SP is the first sample s in the other signal section S3. The sample point SPk' is a positive value, and The next sampling point sp is negative, so it means that the signal segment S3 has completed half the cycle.

Page 14 200410508 V. Description of the invention (11) (ha 1 f per i od). Similarly, when the sampling value of the sampling point SP is negative, and the sampling value of the next sampling point SP is positive, so It means that the signal section S3 has completed one cycle. As mentioned above, the sampling point SPh is the last sampling point in the signal section S 3, so the sampling point corresponding to a certain signal section is obtained through the conventional zero-crossing calculation process (step 106), Since this embodiment deals with a pair of sampling points one by one, when the detection of a signal segment has not been completed in step 106, for example, processing of the sampling point Sp m, since the sampling point SP is the last sample in the non-signaling segment S3 Point SPi i, so the next step 11 2 will be performed. Step 1 12 is used to determine whether the operation of the digital signal processor 34 to detect the audio signal has reached a frame time, such as 1 50ms. When the inspection time is reached, the process returns to step 102 to continue processing the next sampling point SP ffl +. In addition, if a signal segment is judged when executing step 10, it will further judge whether the sampling point of the signal segment is valid (step 108), and step 108 is based on the sampling of the signal segment. The signal energy (signal energy) corresponding to the point and the number of sampling points in the signal section are used as the basis for judgment. The principle is described below. According to the specifications of the conventional continuous audio coding mute system (speci f icat ion), the signal between the audio signal (frequency between 62 · 5Hz ~ 2 50Hz) and the communication signal (frequency between 300Hz ~ 3.4KHz) The energy ratio is in a relationship of 1: 4 ~ 1: 5, so by calculating the signal energy corresponding to the sampling points in the signal section, the noise that does not belong to the audio signal although the frequency is between 62 · 5Hz ~ 2 50Hz Filter out, for example, as in step 102, a signal with a frequency range of 238 ~ 250 Hz is screened out. However, noise from non-audio signals may pass through the frequency.

Page 15 200410508 V. Description of the invention (12) The band in the range of 238 ~ 25 0Hz is filtered through step 10, but the noise may be too weak (for example, the signal generated by the surrounding environment cannot meet the above signal) The condition of the energy ratio, so that the signal section can be skipped through step 108 without performing the subsequent step 11 to record the number of sampling points in the signal section. Therefore, this embodiment sets a predetermined energy value. If the signal energy corresponding to the sampling point of the signal section is lower than the predetermined energy value, ignore the sampling point of the signal section and execute step 11 2 to determine whether the inspection time has been reached. In addition, step i 08 Check the sampling point of the signal section to filter out the signal section that is severely affected by noise. For example, for audio signal T3, the frequency is = ζ '5ΐ = / Digital converter 32 The sampling frequency used is 64 sampling points, which consumes a lot of time: ':-The signal segment should include noise interference _ ,: and! 10 ^ 丄 Smooth filtering processing to attenuate ^ I, ^ ^; Noise makes the tone One missing signal

The second half of the calculation, or the phenomenon that the audio signal is distorted due to the interference of its own circuit during the digital signal processor 3 ’s generation process, so the signal section will temporarily change its cycle: The number of samples corresponding to mί ϊ t will greatly deviate from the ideal value μ, and the number of sampling points in the section and the ideal search will be omitted. The signal section will be omitted, and step 112 will determine whether the Check the time. In other words, if the sampling points of the bluffing section meet the above-mentioned signal energy ratio and the number of samples

fc SH p. 16 200410508

: Only the condition will be established. The number of samples will be advanced! ^ Π10 is used to record the signal section and is used to determine the audio signal. At half-track t, the operation of the bit number processor 34 reaches the inspection time A 9 Μ Η, and the inspection time of the right W Hai is 150 mS. For an audio signal ί r! It ' Article ff 'Also, that is, when step 1 14 starts to be executed, the digital signal J should record the number of samples of 37 cycles. Please note that in this embodiment, only the signal area with a complete cycle is recorded. Segment, the example of the non-complete cycle signal segment that may exist at the beginning of operation is obtained through step 1 1 4 to obtain the sample generated offset (0ffset) recorded in step 1 丨 0, so this embodiment is based on a The effect of the above frequency offset on the number of sampling points is averagely low, so the value can be used to make the digital signal processor 3 4 detect the audio signal later, and calculate the average value and a theoretical value (for the audio signal T 3 rain). Offset (step 11 6) 'Finally, compare the offset with an E to determine whether the voice signal D1 uses the logical channel. If the deviation is less than the threshold, it indicates that the voice signal A1 D1 and the portable wireless pair logic Channel is the same ' The circuit 40 is informed of the relevant operation to output the call of the caller (frequency range 300 Hz ') to the portable radio. On the other hand, if the offset is greater than that, the voice signal Al, D1 and the portable radio are included. The average value is taken as the signal is transmitted between the caller and the receiver to the caller / receiver's movement, and the building is blocked. The T 38 (frequency ^ 37 processors 3 will be executed later) 4 Check the time and ignore the sample point. The process of the number of points will be affected by the frequency production operation. The average is accurate, but it is 64. ^ The threshold can be started by shifting the amount less than the phone. 3. 4KHz) 该Logic for critical value 200410508

The channels are not the same, that is, the digital μ D1 China debt has detected that the portable wireless pair 4 2 = 34 cannot be notified by the voice signal Ab control circuit 40 and then the audio signal used by the household, so the signal processor 34 will re-receive the voice signal A1, and then count 102) to continue the operation of the audio signal of the mobile phone (step ^ T to process the signal received by the wireless walkie-talkie ^ * Si) i This embodiment The disclosed smoothing process is based on the effect of the average operation. However, other methods can also be used to achieve the effect of the second principle. For example, for the sample point to be processed, I, consider the J temple. The processing sampling point is continuous in front of it-a predetermined number (for example) of processed sampling points, and then the two sampling points with = large sampling value and minimum sampling value are removed from the 8 sampling points, and the remaining 6 sampling points are taken Selecting an intermediate value from the sampling values corresponding to the points, or calculating the average of the 6 taken ladder values as the sampling value of the sampling point to be processed, can make / ^ 06 produce a more accurate result. In addition, In this embodiment, when the digital number processor 34 determines After the operation of interrupting the audio signal is started (step 丨 〇〇) is ^ ^ Next step 102, and operates according to the process shown in Figure 4. However, if the digital signal processor 34 determines that the operation of the audio signal is started, (Step '1 00') Start from step 1 12 and follow the procedure shown in Figure 4 to get you T: to reach the point of the present invention. In addition, step 10 6 disclosed in this embodiment corresponds to the voice A period of the signal A 1 is used as a signal section, and a half-period speech signal A1 can also be used as the signal section.

200410508 V. The invention explains that the signal section of the calculation unit 11 8 is used only for the interference that is used to detect the weather alarm and the operation is compared with the analog communication wave processing, to further perform a rate offset pair. Not only can the hardware be reduced. Above} Patent Scope / Scope. (15) The number of samples can be determined by the operation of the zero-crossing rate to determine the start and end of each, and through steps 1 1 4, 11 6 and 'J of the present invention, the purpose of the audio test, Moreover, this embodiment is not familiar with the continuous audio coding mute system, and other devices can also be used to measure a signal of a predetermined frequency, such as a weather alert signal used in North America, whose frequency is ι〇5〇Ηζ, the superior The scope of the invention. According to the conventional technology, the signal is converted to one to obtain a smoother meanwhile, the average number of samples of the filtering filter is more costly. The conventional filtering unit uses signal calculations to reduce the accuracy of the present invention. The invention detects digital communication ratio filters to reduce the proportion of energy. Finally, the signal is low, so the detection result is sufficient for audio signals. The method of taking the wireless transmission of the present invention, and the signal method is to pass through a band to filter the effect of the number of zero-crossing rate samples such as the number of five-segment five-segment input process accompanied by the detection of audio signals is easy to implement digital filtering Sure enough, the number of samples in the processing system varies greatly with the frequency. 200410508 The diagram is a simple illustration. The diagram is simple. Figure 1 is a schematic diagram of the frequency range used in the conventional continuous audio coding mute system. FIG. 2 is a functional block diagram of a conventional portable radio. FIG. 3 is a functional block diagram of the decoder of the present invention. FIG. 4 is an operation flowchart of the digital signal processor shown in FIG. 3. Figure 5 is a waveform diagram of the analog voice signal and the digital voice signal shown in Figure 3. Explanation of Symbols 10 Portable Wireless Interphone 11 Antenna 12 Transceiver 14 Selector 16 ^ 30 Decoder 18 Encoder 20 Voice Signal Processing Unit 22 Mouth Spike 8 24 Microphone 26 Control Unit 32 Analog / Digital Converter 34 Digital Signal Processing Device 36 random access memory 38 read-only memory

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Claims (1)

ZUU41U ^ U «VI. Patent application scope 1. A type of wireless communication attack detection frequency) of the audio signal should be an audio frequency (tone receives an analog voice signal, the method includes: frequency) the analogy 4 The fans use a sampling frequency (sampling the digital voice signal is converted from a complex number A to a digital voice signal, corresponding to a period (pe Γ 丨 〇111 machine number segment, each signal segment at a detection time (Fra ^; and to judge the analogue A-frequency signal according to each number; within number 9 corresponding to a predetermined frequency remainder ΐime), select a plurality of first signal sections in the digital voice signal J;以 信号 区 # ό twist a gas, does the corresponding sample point (°) include the sound corresponding to the audio frequency, where the audio frequency is "located in the predetermined frequency range. For example, if a patent application is filed in each A signal area = the method described in item 1, which further includes: points, and calculates the pre-selection of a predetermined number of samples continuously using the average value of f1 sampling points The average of the values, and the sampling value. The first sampling point of the new predetermined number of sampling points, such as the patent application ΛΛ * point, is the method described in item 2 of the predetermined number of targets, where the first sampling is the last A sampling point. The target sampling point is in the time domain. 200410508 6. Scope of patent application 4. The method described in item 1 of the scope of patent application further includes: detecting the zero-crossing rate (zer 〇- cr 〇ssing rate) corresponding to each first signal segment to determine Its period, and calculate the number of sampling points corresponding to each first signal segment within the detection time. 5. The method according to item 1 of the scope of patent application, further comprising: comparing a predetermined energy value and a signal energy corresponding to each first signal segment (signa 1 energy) to determine whether to select the first signal Section. 6. The method according to item 5 of the scope of patent application, further comprising: calculating an offset between the signal energy corresponding to each first signal segment and the predetermined energy value, if the offset is less than a critical value Value selects the first signal segment. 7. The method according to item 1 of the scope of patent application, further comprising: comparing a predetermined count value and the number of sampling points corresponding to each first signal section to determine whether to select the first signal section. 8. The method as described in item 7 of the scope of patent application, further comprising: calculating an offset between the number of sampling points corresponding to each first signal segment and the predetermined count value, if the offset is less than A threshold value selects the first signal section. , Page 22 200410508 VI. Patent application scope 9 · The method described in item 1 of the patent application scope further includes: calculating an average of the number of sampling points corresponding to the plurality of first signal sections; and comparing The average value and the audio signal are based on the number of sampling points corresponding to the sampling frequency to determine whether the analog voice signal includes the audio signal. I. The method as described in item 9 of the scope of patent application, further comprising: calculating an offset between the number of sampling points corresponding to the audio signal according to the sampling frequency and the average, if the offset is less than A threshold value indicates that the analog voice signal includes the audio signal. II. The method according to item 1 of the scope of patent application, wherein the wireless communication device is a walkie-talki. 1 2 · The method described in item 1 of the scope of patent application, which is applied to a continuous tone-coded squelch system (CTCSS), wherein the wireless communication device is based on an audio signal corresponding to an audio frequency It is different from the signal transmitted on a communication channel. Page 23