WO2016019622A1 - 一种通话过程中信息的传送方法、获取方法、装置及终端 - Google Patents

一种通话过程中信息的传送方法、获取方法、装置及终端 Download PDF

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Publication number
WO2016019622A1
WO2016019622A1 PCT/CN2014/087648 CN2014087648W WO2016019622A1 WO 2016019622 A1 WO2016019622 A1 WO 2016019622A1 CN 2014087648 W CN2014087648 W CN 2014087648W WO 2016019622 A1 WO2016019622 A1 WO 2016019622A1
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call
information
ascii
during
frequency
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PCT/CN2014/087648
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English (en)
French (fr)
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魏国华
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中兴通讯股份有限公司
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Publication of WO2016019622A1 publication Critical patent/WO2016019622A1/zh

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/725Cordless telephones

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  • the present invention relates to the field of communications technologies, and in particular, to a method, an acquisition method, a device, and a terminal for transmitting information during a call.
  • An object of the present invention is to provide a method for transmitting information during a call, an acquisition method, and Devices and terminals enable people to transmit and retrieve in real time, quickly and accurately when they need to transmit information during a call.
  • an embodiment of the present invention provides a method for transmitting information during a call, including:
  • the voice signal is mixed into a call voice stream and sent to the recipient communication terminal.
  • the binary sequence of the ASCII code corresponding to the information to be sent during the call is obtained, including:
  • the step of obtaining the binary sequence of the ASCII code corresponding to the information to be sent during the call includes:
  • the binary sequence of the ASCII code is converted into a voice signal according to the frequency and time of the preset binary number, including:
  • the embodiment of the invention further provides a method for acquiring information during a call, which includes:
  • the voice signal is parsed to obtain information to be sent during the call and displayed on the call interface of the receiving communication terminal.
  • the obtaining method further includes:
  • the information to be sent during the call is saved by calling the address book interface of the receiving communication terminal.
  • the voice signal is parsed to obtain information to be sent during the call and displayed on the call interface of the receiving communication terminal, including:
  • the information to be sent during the call is displayed on the call interface of the receiving communication terminal.
  • the embodiment of the invention further provides a device for transmitting information during a call, comprising:
  • Obtaining a module configured to obtain a binary sequence of ASCII codes corresponding to information to be sent during the call;
  • a conversion module configured to convert the binary sequence of the ASCII code into a voice signal according to a frequency and a time of a preset binary number
  • the sending module is configured to mix the voice signal into the call voice stream and send the signal to the receiver communication terminal.
  • the obtaining module includes:
  • the first obtaining submodule is configured to sequentially input the information to be sent during the call by using a soft keyboard of the sending communication terminal;
  • a second obtaining submodule configured to obtain, according to an ASCII code table, a decimal code of an ASCII code corresponding to the information to be sent during the call;
  • a third obtaining submodule configured to convert the decimal code into a binary sequence of corresponding ASCII codes.
  • the obtaining module further includes:
  • the fourth obtaining submodule is configured to invoke an address book interface of the sending communication terminal to obtain information to be sent during the call;
  • a fifth obtaining submodule configured to respond to the transmitting operation of the information to be sent during the call, and convert the information to be sent during the call into a decimal code corresponding to the ASCII code according to the ASCII table;
  • a sixth obtaining submodule configured to convert the decimal code into a binary sequence of corresponding ASCII codes.
  • the conversion module includes:
  • a preset module configured to preset a frequency of a binary number 1 and a binary number 0, respectively, and a duration of the preset frequency of the binary number 1 and the binary number 0;
  • the generating module is configured to process the binary sequence of the ASCII code by using a Goertzel basic algorithm, and perform tone detection at a preset position to generate a voice signal including an ASCII code sequence.
  • An embodiment of the present invention further provides an apparatus for acquiring information during a call, including:
  • a voice acquisition module configured to acquire a voice signal including an ASCII code sequence
  • the parsing module is configured to parse the voice signal to obtain information to be sent during the call and display on the call interface of the receiving communication terminal.
  • the obtaining device further includes:
  • the saving module is configured to save the information to be sent during the call by calling the address book interface of the receiving communication terminal.
  • the parsing module includes:
  • a first parsing submodule configured to separately acquire a horizontal frequency and a column frequency of the voice signal
  • a second parsing submodule configured to obtain a corresponding binary code sequence according to a horizontal frequency and a column frequency of the speech signal
  • a third parsing sub-module configured to compare the frequency and amplitude of the horizontal and vertical frequencies with a preset range, and determine the binary code when the frequency and the amplitude are both within the preset range
  • the sequence is an ASCII code sequence
  • a fourth parsing sub-module configured to obtain, according to the ASCII code sequence, information to be sent during a corresponding call
  • the fifth parsing sub-module is configured to display the information to be sent during the call on the call interface of the receiving communication terminal.
  • the embodiment of the present invention further provides a terminal, including the information transmission device during the call as described above, and the information acquisition device during the call as described above.
  • Embodiments of the present invention also provide a computer storage medium having stored therein computer executable instructions for performing the above method.
  • the receiver analyzes the above voice signal and displays the information to be sent during the call, which realizes the convenience of transmitting the information and greatly enhances the user experience; and adopts the method of transmitting ASCII code at the same time. It has the characteristics of high security and encryption authentication, and has a good application prospect.
  • FIG. 1 is a schematic diagram showing the basic steps of a method for transmitting information during a call according to an embodiment of the present invention
  • FIG. 2 is a flowchart showing a specific embodiment of a method for transmitting information during a call according to an embodiment of the present invention
  • FIG. 3 is a schematic diagram showing the basic steps of a method for acquiring information during a call according to an embodiment of the present invention
  • FIG. 4 is a schematic diagram of an information parsing process in a method for acquiring information during a call according to an embodiment of the present invention
  • FIG. 5 is a flowchart of a specific embodiment of a method for acquiring information during a call according to an embodiment of the present invention
  • FIG. 6 is a schematic structural diagram of an apparatus for transmitting information during a call according to an embodiment of the present invention.
  • FIG. 7 is a schematic structural diagram of an apparatus for acquiring information during a call according to an embodiment of the present invention.
  • the present invention is directed to the problem that the information needs to be transmitted by means of short message, WeChat, etc. during the call process in the prior art, but there is a problem of delay; it can also be transmitted by DTMF, but the method only transmits numbers and the security cannot be guaranteed.
  • Providing a method and a method for transmitting information during a call by converting a message to be sent during a call into a binary signal containing a binary sequence of ASCII code, and mixing it into a voice stream for transmission, thereby achieving no need to hang
  • the telephone is disconnected and the information is transmitted in real time; at the same time, the receiving party parses the above voice signal and displays the information to be sent during the call, thereby realizing the convenience of transmitting the information and greatly enhancing the user experience; and the method of transmitting by using ASCII code has High security and encryption authentication have good application prospects.
  • ASCII is a computerized coding system based on the Latin alphabet. It is mainly used to display modern English and other Western European languages. It is the most versatile single-byte encoding system available today and is equivalent to the international standard ISO/IEC 646.
  • all data is stored in binary numbers (because the computer uses high and low levels to represent 1 and 0, respectively), for example, 52 letters like a, b, c, d (including uppercase), and 0, 1 and other numbers and some commonly used symbols (such as *, #, @, etc.) should also be represented by binary numbers when stored in the computer, and which binary numbers are used to represent which symbols, of course People can agree on their own set (this is called coding), and if you want to communicate with each other without causing confusion, then everyone must use the same coding rules, so the American standardization organization has issued ASCII code. The same specifies which binary numbers are used to represent the above common symbols.
  • the American Standard Information Exchange Code is a standard single-byte character encoding scheme developed by the American National Standards Institute (ANSI) for text-based data. It began in the late 1950s and was finalized in 1967. It was originally a US national standard for Western computer character encoding standards used by different computers when communicating with each other. It has been designated as an international standard by the International Organization for Standardization (ISO) and is called the ISO 646 standard. Applies to all Latin alphabet letters.
  • ISO International Organization for Standardization
  • ASCII uses a specified combination of 7-bit or 8-bit binary numbers to represent 128 or 256 possible characters.
  • Standard ASCII codes are also called basic ASCII codes, using 7-bit binary numbers to represent all uppercase and lowercase letters, numeric words 0 through 9, punctuation, and special control characters used in American English. among them:
  • control characters or communication-specific characters are control characters or communication-specific characters (the rest are displayable characters), such as control characters: LF (line feed), CR (carriage return), FF (page change), DEL (delete) , BS (backspace), BEL (bell), etc.; communication-specific characters: SOH (text header), EOT (text end), ACK (confirmation), etc.; ASCII values of 8, 9, 10, and 13 are converted to retreat Grid, tabulation, line feed, and carriage return characters. They don't have a specific graphical display, but they have different effects on text display depending on the application.
  • 32 to 126 are characters (32 is a space), of which 48 to 57 are 10 to 9 Arabic numerals; 65 to 90 are 26 uppercase English letters, 97 to 122 It is 26 lowercase English letters, and the rest are some punctuation marks, arithmetic symbols, and so on.
  • parity check is a method used to check whether an error occurs during code transmission. It is generally divided into odd parity and even parity. Odd parity rule: the correct code must be an odd number of one byte, if not an odd number, then add 1 to the highest bit b7; even parity specifies: the correct code, the number of 1 in a byte must be even If it is not even, add 1 to the highest bit b7.
  • Extended ASCII codes The last 128 are called extended ASCII codes.
  • Extended (or "high") ASCII is supported on many x86 based systems. Extended ASCII allows the 8th bit of each character to be used to determine the additional 128 special symbol characters, foreign letters, and graphic symbols.
  • the embodiment of the invention provides a method for transmitting information during a call, as shown in FIG. 1 , which includes:
  • Step 1 Obtain a binary sequence of ASCII codes corresponding to information to be sent during a call
  • Step 2 Convert the binary sequence of the ASCII code into a voice signal according to a frequency and a time of the preset binary number
  • step 3 the voice signal is mixed into a call voice stream and sent to the receiver communication terminal.
  • the ASCII code is a set of computer coding systems based on the Latin alphabet. It is mainly used to display modern English and other Western European languages. It is the most versatile single-byte encoding system available today and is equivalent to the international standard ISO/IEC 646.
  • the information to be sent during the call mainly includes the contact information of the third party, such as the telephone number of the third party, the email address, and the like. As shown in Table 1, in the specific embodiment of the present invention, fields of 0-31 and 127 of ASCII code are not currently used, and uppercase and lowercase letters, numbers, and special symbols of other fields can satisfy the naming of the email address and the telephone number.
  • step 1 includes:
  • Step 11 sequentially input the information to be sent during the call by using a soft keyboard of the sender communication terminal;
  • Step 12 Acquire, according to the ASCII code table, a decimal code of an ASCII code corresponding to the information to be sent during the call;
  • step 13 the decimal code is converted into a binary sequence of corresponding ASCII codes.
  • the information to be sent during the call is the telephone number 13314528888, and the ASCII code table of Table 1 is looked up, the corresponding ASCII code is decimal 49, 51, 51, 49, 52, 53, 50, 56. , 56, 56, 56; then the binary is: 110001, 110011, 110011, 110001, 110100, 110101, 110010, 111000, 111000, 111000, 111000.
  • one method is that if the user has already remembered the required phone number or email address, only A soft keyboard similar to the input method (because the general call menu only has a dial pad, only the number keys, which cannot meet the requirements of sending all characters and numbers in the present application), the phone number or email address is entered in order, then the phone number is Or the email address can be converted into a voice signal followed by the voice stream to be sent to the receiver, and the receiver can also accurately receive the contact information sent by the user.
  • the soft keyboard refers to a QWERTY keyboard (full keyboard).
  • step 1 further includes:
  • Step 14 invoke an address book interface of the sender communication terminal to obtain information to be sent during the call;
  • Step 15 In response to the transmission operation of the information to be sent during the call, convert the information to be sent during the call into a decimal code corresponding to the ASCII code according to the ASCII table;
  • step 16 the decimal code is converted into a binary sequence of corresponding ASCII codes.
  • step 14 is executed to invoke the sender communication terminal.
  • the address book interface query the information to be sent during the corresponding call, that is, the contact information, phone number or email address of the third party.
  • the transfer operation may be to select a contact to be sent in the contact directory, operate the menu option, and click “send via ASCII”, ie
  • the contact information can be sent to the other party through the voice channel, ie, the CS domain of the cell station.
  • step 16 is performed to convert the decimal code into a binary sequence of the corresponding ASCII code.
  • step 2 includes:
  • Step 21 preset the frequency of the binary number 1 and the binary number 0, respectively, and the duration of the preset frequency of the binary number 1 and the binary number 0;
  • step 22 the binary sequence of the ASCII code is processed by a Goertzel basic algorithm, and tone detection is performed at a preset position to generate a voice signal including an ASCII code sequence.
  • the basic principle of the algorithm for generating a speech signal is as follows: a frequency representing a binary number of 1 is set to a high frequency of 1800 Hz, and a frequency representing a binary number of 0 is set to 1400 Hz.
  • the frequency representing 1 or 0 respectively needs a duration of 22 ms.
  • the amplitude and time requirements are specified and limited by the specific embodiment of the present invention, and are closely related to the core algorithm. The parsing and generation are all related to this. It can avoid unnecessary erroneous detection caused by interference of normal speech signals.
  • each character 1 is sent as a start bit, and 0 is a stop bit. Then the frequency of sending a number or character during the call is about 154ms.
  • the Goertzel basic algorithm can be used to derive the same real and imaginary parts of the frequency as a conventional discrete Fourier transform (DFT) or (fast Fourier transform) FFT. Amplitude and phase information can also be calculated from the real and imaginary parts of the frequency, if desired.
  • the Goertzel basic algorithm is processed immediately after each sample, and a tone detection is performed every Nth sample.
  • a tone detection is performed every Nth sample.
  • the signal generated by the Goertzel basic algorithm is analog in the time domain, the detection thereof needs to be sampled to form a digital signal.
  • the maximum frequency is 1800 Hz.
  • the 1400HZ and 1800HZ frequencies required by this algorithm are all within the human audible range.
  • the sampling rate of 8 kHz is commonly used in telecommunication applications. That is, 8000 samples per second, according to Nyquist's sampling law, can fully meet the requirements.
  • step 101 both parties establish a call
  • Step 102 the call party A requests the call party B to send the contact information (phone number and email address) of the contact person A;
  • Step 103 Determine whether the call party B remembers the contact mode of the contact A;
  • Step 104 if the caller B remembers the contact mode of the contact A, directly invokes a special soft keyboard in the menu, and inputs the contact mode number in the order of the soft keyboard;
  • Step 105 If the party B does not remember the contact mode of the contact A, then the party address interface of the party B is called, and the corresponding contact A is queried, and then the “contact mode is sent by ASCII code” is selected in the menu, and the contact A is The contact method will perform key operation on the contact mode in the background through the built-in ASCII code encoding program to simulate the keyboard input mode;
  • Step 106 Convert the ASCII code frequency signal representing the character and value of the contact mode into a voice signal and mix it into the sending direction and put it into the PCM voice stream and send it to the calling party.
  • the embodiment of the present invention further provides a method for acquiring information during a call, including:
  • Step 4 Acquire a voice signal including an ASCII code sequence
  • step 5 the voice signal is parsed to obtain information to be sent during the call and displayed on the call interface of the receiving communication terminal.
  • the acquiring method further includes:
  • Step 6 Save the information to be sent during the call by calling the address book interface of the receiving communication terminal.
  • the receiving communication terminal when it correctly acquires the handshake signal of the contact mode sent by the opposite end, it starts to detect the information in the PCM stream of the voice in real time, when the PCM voice in the voice stream contains numbers or letters.
  • the number information such as 0-9, *, #, ., @, AZ, az, etc. is extracted.
  • the interruption is generated, and the user is given a complete prompt, and the user can save the contact phone number and the mobile phone number in real time during the call. This way of interrupting greatly saves system resources and power consumption compared to the round-robin approach.
  • step 5 includes:
  • Step 51 respectively acquiring a line frequency and a column frequency of the voice signal
  • Step 52 Acquire a corresponding binary code sequence according to a horizontal frequency and a column frequency of the voice signal.
  • Step 53 Compare the frequency and amplitude of the horizontal frequency and the serial frequency with a preset range, and determine that the binary code sequence is an ASCII code sequence when the frequency and the amplitude are both within the preset range. ;
  • Step 54 Obtain, according to the ASCII code sequence, information to be sent during a corresponding call;
  • Step 55 Display the information to be sent during the call on the call interface of the receiving communication terminal.
  • the DSP detects whether the peer sends an ASCII code signal in real time.
  • the party successfully receives the voice containing the ASCII code audio sequence the built-in call is immediately invoked.
  • the ASCII parser parses the speech.
  • the built-in ASCII parser looks for the horizontal and vertical frequencies of the signal, only the frequency and amplitude of the algorithm are full. It is the correct ASCII code that meets the requirements, which effectively filters out harmonic interference in some voices.
  • the result of the final analysis is a continuous value of 0-9 or other characters such as A-Z, a-z, *, @, #, etc., and then prompts to the party side in the call interface, and calls the address book interface to prompt to save or give up.
  • Step 201 The calling party A receives the voice data including the ASCII code sequence
  • Step 202 Invoking the DSP internal decoding program of the calling party to parse the ASCII code data
  • Step 203 displaying the completed complete phone number or email address on the call interface of the calling party A;
  • Step 204 Calling the address book interface of the calling party to save the contact information
  • step 205 the calling party A successfully saves the contact information of the resolved contact A.
  • the corresponding ASCII code is 49, 51, 51, 49, 52, 53, 50, 56, 56, 56, 56, 56, and the binary is: 110001, 110011, 110011, 110001 , 110100, 110101, 110010, 111000, 111000, 111000, 111000.
  • the corresponding ASCII code is 119, 103, 104, 64, 122, 116, 101, 46, 99, 111, 109, 46, 99, 110.
  • the binary is 1110111 (w), 1100111 (g), 1101000 (h), 1000000 (@), 1111010 (z), 1110100 (t), 1100101 (e), 101110 (.), 1100011 (c), 1101111 ( o), 1101101(m), 101110(.), 1100011(c), 1101110(n).
  • the ASCII code transmitting module will generate an audio signal according to the frequency and time requirements of the algorithm according to the binary sequence representing the contact mode, and mix it into the uplink call voice stream.
  • the detection algorithm of the ASCII code parsing module is turned on, The voice stream of the line is detected in real time. Firstly, the frequency and time sent by the sender are parsed to satisfy the ASCII code, which character and number are selected according to the ASCII code, and other special characters, and then the final received contact is determined according to the combination. Ways such as phone number and email address. If the phone number and the email address are transmitted at the same time, the algorithm will be sent in the prescribed order, and a blank content is preset as a dividing line between the two contents to correctly distinguish the phone number and the email address.
  • the embodiment of the present invention further provides a device for transmitting information during a call, including:
  • the obtaining module 10 is configured to acquire a binary sequence of ASCII codes corresponding to information to be sent during the session;
  • the conversion module 20 is configured to convert the binary sequence of the ASCII code into a voice signal according to a preset frequency and time of the binary number;
  • the sending module 30 is configured to mix the voice signal into the call voice stream and send the signal to the receiver communication terminal.
  • the acquiring module 10 includes:
  • the first obtaining submodule is configured to sequentially input the information to be sent during the call by using a soft keyboard of the sending communication terminal;
  • a second obtaining submodule configured to obtain, according to an ASCII code table, a decimal code of an ASCII code corresponding to the information to be sent during the call;
  • a third obtaining submodule configured to convert the decimal code into a binary sequence of corresponding ASCII codes.
  • the acquiring module 10 further includes:
  • the fourth obtaining submodule is configured to invoke an address book interface of the sending communication terminal to obtain information to be sent during the call;
  • a fifth acquisition submodule configured to respond to the transmission operation of the information to be sent during the call Performing, according to the ASCII table, converting the information to be sent during the call into a decimal code corresponding to the ASCII code;
  • a sixth obtaining submodule configured to convert the decimal code into a binary sequence of corresponding ASCII codes.
  • the conversion module 20 includes:
  • a preset module configured to preset a frequency of a binary number 1 and a binary number 0, respectively, and a duration of the preset frequency of the binary number 1 and the binary number 0;
  • the generating module is configured to process the binary sequence of the ASCII code by using a Goertzel basic algorithm, and perform tone detection at a preset position to generate a voice signal including an ASCII code sequence.
  • the transmitting device provided by the embodiment of the present invention is a device applying the foregoing transmitting method, and all embodiments of the foregoing transmitting method are applicable to the device, and all of the same or similar beneficial effects can be achieved.
  • the embodiment of the present invention further provides an apparatus for acquiring information during a call, including:
  • the voice acquiring module 40 is configured to acquire a voice signal including an ASCII code sequence
  • the parsing module 50 is configured to parse the voice signal to obtain information to be sent during the call and display it on the call interface of the receiving communication terminal.
  • the acquiring device further includes:
  • the saving module 60 is configured to save the information to be sent during the call by calling the address book interface of the receiving communication terminal.
  • the parsing module 50 includes:
  • a first parsing submodule configured to separately acquire a horizontal frequency and a column frequency of the voice signal
  • a second parsing submodule configured to obtain a corresponding binary code sequence according to a horizontal frequency and a column frequency of the speech signal
  • a third parsing sub-module configured to compare the frequency and amplitude of the horizontal and vertical frequencies with a preset range, and determine the binary code when the frequency and the amplitude are both within the preset range
  • the sequence is an ASCII code sequence
  • a fourth parsing sub-module configured to obtain, according to the ASCII code sequence, information to be sent during a corresponding call
  • the fifth parsing sub-module is configured to display the information to be sent during the call on the call interface of the receiving communication terminal.
  • the receiving party parses the voice signal and displays the information to be sent during the call, thereby realizing the convenience of transmitting the information, greatly enhancing the user experience, and transmitting the ASCII code at the same time.
  • the method has the characteristics of high security and encryption authentication, and has a good application prospect.
  • the acquiring apparatus is the apparatus applying the foregoing obtaining method, and all the embodiments of the foregoing obtaining method are applicable to the obtaining apparatus, and all of the same or similar beneficial effects can be achieved.
  • the embodiment of the present invention further provides a terminal, including the information transmission device during the call as described above, and the information acquisition device during the call as described above.
  • the transmitting device and the obtaining device in the specific embodiment of the present invention need to be used in pairs to complete the complete function of sending and parsing, and the single party cannot complete the overall function.
  • the patented invention has the characteristics of high security, encryption authentication and the like, and will be more and more widely used in the application of smart terminals in the future.
  • the terminal provided by the embodiment of the present invention is a terminal including the foregoing transmitting device and the foregoing acquiring device, and all embodiments of the transmitting device and the obtaining device and the beneficial effects thereof are applicable to the terminal.
  • the embodiment of the present invention further provides a computer storage medium, wherein computer executable instructions are stored, and the computer executable instructions are used to execute the foregoing method.
  • Each of the above modules may be implemented by a central processing unit (CPU), a digital signal processor (DSP), or a field-programmable gate array (FPGA) in the electronic device.
  • CPU central processing unit
  • DSP digital signal processor
  • FPGA field-programmable gate array
  • embodiments of the present invention can be provided as a method, system, or computer program product. Accordingly, the present invention can take the form of a hardware embodiment, a software embodiment, or a combination of software and hardware. Moreover, the invention can take the form of a computer program product embodied on one or more computer-usable storage media (including but not limited to disk storage and optical storage, etc.) including computer usable program code.
  • the computer program instructions can also be stored in a computer readable memory that can direct a computer or other programmable data processing device to operate in a particular manner, such that the instructions stored in the computer readable memory produce an article of manufacture comprising the instruction device.
  • the apparatus implements the functions specified in one or more blocks of a flow or a flow and/or block diagram of the flowchart.

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Abstract

本发明提供一种通话过程中信息的传送方法、获取方法、装置及终端,涉及通信技术领域。其中,该传送方法包括:获取通话过程中要发送的信息对应的ASCII码的二进制序列;根据预设的二进制数的频率和时间,将ASCII码的二进制序列转化成语音信号;将所述语音信号混音到通话语音流中发送给接收方通讯终端。

Description

一种通话过程中信息的传送方法、获取方法、装置及终端 技术领域
本发明涉及通信技术领域,特别涉及一种通话过程中信息的传送方法、获取方法、装置及终端。
背景技术
在目前的移动终端上,当两个用户建立通话后,其中一方想知道对方熟悉的朋友的联系方式的话,如果对方能够记住联系方式的话,这时候用户只能将通话应用程序挂起到后台,然后用短消息的传统方式,或者使用QQ,微信等数据应用来发送,比较费时费力费钱。这些传统方式大多存在一定的网络延时,且运营商需要根据短消息个数或数据流量来收取费用。另外当对方想要找到你刚发的联系人时,还得查询短消息应用或者其他应用,如果这些应用有比较好的用户体验,添加了按照电话号码或者手机号码来进行联系人添加,还相对方便,如果没有这些功能,对方不得不通过记忆或者找来纸和笔来记录,再在电话薄或通讯录中去添加,这样极大的浪费了时间。虽然现在有一些技术能够通过双音多频DTMF方式在通话中传递电话号码,但是这种技术目前仅限于传递0-9的数字,而无法传送油箱地址,油箱在现在科技社会中所起的作用毋庸置疑,所以通话中传递油箱地址也变得越来越迫切。另外DTMF作为移动电话和固定电话的标准通用协议,一些不法份子经常通过非法途径获取通话录音并解析DTMF信息,非法获取用户的密码等安全信息,目前安全性无法保障。
发明内容
本发明的目的在于提供一种通话过程中信息的传送方法、获取方法、 装置及终端,让人们在通话过程中需要传送信息时,能够实时、快捷、准确的传送和获取。
为了达到上述目的,本发明实施例提供一种通话过程中信息的传送方法,包括:
获取通话过程中要发送的信息对应的ASCII码的二进制序列;
根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
将所述语音信号混音到通话语音流中发送给接收方通讯终端。
其中,所述获取通话过程中要发送的信息对应的ASCII码的二进制序列,包括:
通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
将所述十进制码转化为对应的ASCII码的二进制序列。
其中,所述获取通话过程中要发送的信息对应的ASCII码的二进制序列的步骤还包括:
调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
响应所述通话过程中要发送的信息的传送操作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
将所述十进制码转化为对应的ASCII码的二进制序列。
其中,根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号,包括:
分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在 预设位置进行音调检测,生成包含ASCII码序列的语音信号。
本发明实施例还提供一种通话过程中信息的获取方法,包括:
获取包含ASCII码序列的语音信号;
解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
其中,所述获取方法还包括:
通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
其中,解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示,包括:
分别获取所述语音信号的行频和列频;
根据所述语音信号的行频和列频,获取对应的二进制码序列;
分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
本发明实施例还提供一种通话过程中信息的传送装置,包括:
获取模块,配置为获取第通话过程中要发送的信息对应的ASCII码的二进制序列;
转换模块,配置为根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
发送模块,配置为将所述语音信号混音到通话语音流中发送给接收方通讯终端。
其中,所述获取模块包括:
第一获取子模块,配置为通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
第二获取子模块,配置为根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
第三获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
其中,所述获取模块还包括:
第四获取子模块,配置为调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
第五获取子模块,配置为响应所述通话过程中要发送的信息的传送操作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
第六获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
其中,所述转换模块包括:
预设模块,配置为分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
生成模块,配置为利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在预设位置进行音调检测,生成包含ASCII码序列的语音信号。
本发明实施例还提供一种通话过程中信息的获取装置,包括:
语音获取模块,配置为获取包含ASCII码序列的语音信号;
解析模块,配置为解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
其中,所述获取装置还包括:
保存模块,配置为通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
其中,所述解析模块包括:
第一解析子模块,配置为分别获取所述语音信号的行频和列频;
第二解析子模块,配置为根据所述语音信号的行频和列频,获取对应的二进制码序列;
第三解析子模块,配置为分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
第四解析子模块,配置为根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
第五解析子模块,配置为将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
本发明实施例还提供一种终端,包括如上所述的通话过程中信息的传送装置和如上所述的通话过程中信息的获取装置。
本发明实施例还提供一种计算机存储介质,其中存储有计算机可执行指令,所述计算机可执行指令用于执行上述的方法。
本发明实施例的通话过程中信息的传送方法及获取方法中,通过将通话过程中要发送的信息转化成包含ASCII码的二进制序列的语音信号,并混到通话语音流中发送,实现了无需挂断电话,实时传送信息;同时接受方解析上述语音信号,并显示该通话过程中要发送的信息,实现了传送信息的便捷性,极大的增强了用户体验;同时采用ASCII码传送的方法具有安全性高及加密鉴权性等特性,具有较好的应用前景。
附图说明
图1表示本发明实施例的通话过程中信息的传送方法的基本步骤示意 图;
图2表示本发明实施例的通话过程中信息的传送方法的具体实施例的流程图;
图3表示本发明实施例的通话过程中信息的获取方法的基本步骤示意图;
图4表示本发明实施例的通话过程中信息的获取方法中的信息解析过程示意图;
图5表示本发明实施例的通话过程中信息的获取方法的具体实施例的流程图;
图6表示本发明实施例的通话过程中信息的传送装置的结构示意图;
图7表示本发明实施例的通话过程中信息的获取装置的结构示意图。
具体实施方式
为使本发明要解决的技术问题、技术方案和优点更加清楚,下面将结合附图及具体实施例进行详细描述。
本发明针对现有技术中通话过程中需传送信息时可通过短信、微信等方式传送,但存在延时的问题;也可通过DTMF方式传送,但该方式仅传递数字且安全性无法保障的问题,提供一种通话过程中信息的传送方法及获取方法中,通过将通话过程中要发送的信息转化成包含ASCII码的二进制序列的语音信号,并混到通话语音流中发送,实现了无需挂断电话,实时传送信息;同时接受方解析上述语音信号,并显示该通话过程中要发送的信息,实现了传送信息的便捷性,极大的增强了用户体验;同时采用ASCII码传送的方法具有安全性高及加密鉴权性等特性,具有较好的应用前景。
ASCII码是基于拉丁字母的一套电脑编码系统。它主要用于显示现代英语和其他西欧语言。它是现今最通用的单字节编码系统,并等同与国际标准ISO/IEC 646。
在计算机中,所有的数据在存储和运算时都要使用二进制数表示(因为计算机用高电平和低电平分别表示1和0),例如,像a、b、c、d这样的52个字母(包括大写)、以及0、1等数字还有一些常用的符号(例如*、#、@等)在计算机中存储时也要使用二进制数来表示,而具体用哪些二进制数字表示哪个符号,当然每个人都可以约定自己的一套(这就叫编码),而大家如果要想互相通信而不造成混乱,那么大家就必须使用相同的编码规则,于是美国有关的标准化组织就出台了ASCII编码,同一规定了上述常用符号用哪些二进制数来表示。
美国标准信息交换代码是由美国国家标准学会(American National Standard Institute,ANSI)制定的,标准的单字节字符编码方案,用于基于文本的数据。起始于50年代后期,在1967年定案。它最初是美国国家标准,供不同计算机在相互通信时用作共同遵守的西文字符编码标准,它已被国际标准化组织(International Organization for Standardization,ISO)定为国际标准,称为ISO 646标准。适用于所有拉丁文字字母。
ASCII使用指定的7位或8位二进制数组合来表示128或256种可能的字符。标准ASCII码也叫基础ASCII码,使用7位二进制数来表示所有的大写和小写字母,数字字0到9、标点符号,以及在美式英语中使用的特殊控制字符。其中:
0~31及127(共33个)是控制字符或通信专用字符(其余为可显示字符),如控制符:LF(换行)、CR(回车)、FF(换页)、DEL(删除)、BS(退格)、BEL(响铃)等;通信专用字符:SOH(文头)、EOT(文尾)、ACK(确认)等;ASCII值为8、9、10和13分别转换为退格、制表、换行和回车字符。它们并没有特定的图形显示,但会依不同的应用程序,而对文本显示有不同的影响。32~126(共95个)是字符(32是空格),其中48~57为0到9的十个阿拉伯数字;65~90为26个大写英文字母,97~122号 为26个小写英文字母,其余为一些标点符号、运算符号等。
同时还要注意,在标准ASCII中,其最高位(b7)用作奇偶校验位。所谓奇偶校验,是指在代码传送过程中用来检验是否出现错误的一种方法,一般分为奇校验和偶校验两种。奇校验规定:正确的代码一个字节中1的个数必须是奇数,若非奇数,则在最高位b7添1;偶校验规定:正确的代码一个字节中1的个数必须是偶数,若非偶数,则在最高位b7添1。
后128个称为扩展ASCII码。许多基于x86的系统都支持使用扩展(或“高”)ASCII。扩展ASCII码允许将每个字符的第8位用于确定附加的128个特殊符号字符、外来语字母和图形符号。
常用的ASCII码如表1所示:
ASCII值 控制字符 ASCII值 控制字符 ASCII值 控制字符 ASCII值 控制字符
0 NUT 32 (space) 64 @ 96
1 SOH 33 65 A 97 a
2 STX 34 66 B 98 b
3 ETX 35 # 67 C 99 c
4 EOT 36 $ 68 D 100 d
5 ENQ 37 69 E 101 e
6 ACK 38 & 70 F 102 f
7 BEL 39 , 71 G 103 g
8 BS 40 ( 72 H 104 h
9 HT 41 ) 73 I 105 i
10 LF 42 * 74 J 106 j
11 VT 43 + 75 K 107 k
12 FF 44 , 76 L 108 l
13 CR 45 - 77 M 109 m
14 SO 46 . 78 N 110 n
15 SI 47 / 79 O 111 o
16 DLE 48 0 80 P 112 p
17 DCI 49 1 81 Q 113 q
18 DC2 50 2 82 R 114 r
19 DC3 51 3 83 X 115 s
20 DC4 52 4 84 T 116 t
21 NAK 53 5 85 U 117 u
22 SYN 54 6 86 V 118 v
23 TB 55 7 87 W 119 w
24 CAN 56 8 88 X 120 x
25 EM 57 9 89 Y 121 y
26 SUB 58 : 90 Z 122 z
27 ESC 59 91 [ 123 {
28 FS 60 < 92 / 124 |
29 GS 61 93 ] 125 }
30 RS 62 > 94 ^ 126
31 US 63 95 127 DEL
表1
本发明实施例提供一种通话过程中信息的传送方法,如图1所示,包括:
步骤1,获取通话过程中要发送的信息对应的ASCII码的二进制序列;
步骤2,根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
步骤3,将所述语音信号混音到通话语音流中发送给接收方通讯终端。
本发明一具体实施例中,ASCII码是基于拉丁字母的一套电脑编码系统。它主要用于显示现代英语和其他西欧语言。它是现今最通用的单字节编码系统,并等同于国际标准ISO/IEC 646。本发明的具体实施例中,通话过程中要发送的信息主要包括第三方的联系方式,例如第三方的电话号码、邮箱地址等等。如表1所示,本发明的具体实施例中目前没有使用ASCII码的0-31及127的字段,其他字段的大小写字母、数字以及特殊符号等已经能够满足邮箱地址和电话号码的命名。
可选的,本发明具体实施例中,步骤1包括:
步骤11,通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
步骤12,根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
步骤13,将所述十进制码转化为对应的ASCII码的二进制序列。
本发明一具体实施例中,假如通话中要发送的信息为电话号码13314528888,查找表1的ASCII码表,则对应的ASCII码为十进制49,51,51,49,52,53,50,56,56,56,56;则二进制为:110001,110011,110011,110001,110100,110101,110010,111000,111000,111000,111000。
本发明实施例中,当两个用户建立通话后,其中一方想知道用户熟悉的朋友的联系方式时,一种方法是如果用户已经记住了所需的电话号码或邮箱地址,则仅需在类似输入法的软键盘(因为一般的通话菜单仅仅有拨号盘,仅仅有数字键,不能够满足本申请的发送所有字符和数字的需求)上按照顺序输入电话号码或邮箱地址,则该电话号码或邮箱地址就能够转化为语音信号跟随通话语音流一同发送给接收方,则接收方同样可以准确地接收到用户所发送的联系方式。较佳的,软键盘指QWERTY键盘(全键盘)。
可选的,本发明具体实施例中,步骤1还包括:
步骤14,调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
步骤15,响应所述通话过程中要发送的信息的传送操作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
步骤16,将所述十进制码转化为对应的ASCII码的二进制序列。
本发明一具体实施例中,当两个用户建立通话后,其中一方想知道用户熟悉的朋友的联系方式时,如果发送方没有记住第三方联系方式,则执行步骤14,调用发送方通讯终端的通讯录接口,查询到对应的通话过程中要发送的信息,即第三方的联系方式,电话号码或邮箱地址。具体的,步骤15中的响应所述通话过程中要发送的信息的传送操作,所述传送操作可以为在联系人目录选中要发送的联系人,操作菜单选项,点击“通过ASCII发送“,即可通过语音通道,即小区站CS域,将联系方式发送给对方。
需要说明的是,点击“通过ASCII发送“后,此时通话过程中要发送的信息将通过内置于发送方通讯终端的模拟键盘输入方式在后台对该信息进行了按键操作,将该信息转化为ASCII码的十进制码;再执行步骤16,将所述十进制码转化为对应的ASCII码的二进制序列。
本发明一具体实施例中,步骤2包括:
步骤21,分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
步骤22,利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在预设位置进行音调检测,生成包含ASCII码序列的语音信号。
本发明具体实施例中,生成语音信号使用的算法的基本原理如下:代表二进制数1的频率设定高频1800HZ,代表二进制数0的频率设定为1400HZ。代表1或者0的频率分别需要持续时间为22ms,另外对1400HZ或1800HZ的频率幅值也有一定的要求,幅值和时间的要求是由本发明具体实施例规定和限制的,与核心算法息息相关,频率的解析和生成都与此有关。则可以避免受到正常语音信号的干扰造成不必要的误检测。另外每发送一个字符1为起始位,0为终止位。那么在通话过程中发送一个数字或字符的频率时间大约为154ms。
本发明的具体实施例中,使用Goertzel基本算法能得出与常规离散傅立叶变换(DFT)或(快速傅氏变换)FFT相同的频率实部和虚部。如果需要的话,还可以从该频率实部和虚部中计算出幅度和相位信息。Goertzel基本算法是在每次采样后立即进行处理,在每个第N次采样进行一次音调检测。在采用FFT算法时,我们要对成块的采样进行处理,但这并不意味着必须按块来处理数据。数字处理的时间很短,因此如果每次采样都存在一次中断,那么这些数字处理完全可以在中断服务程序(ISR)内完成。或者,如果系统中存在采样缓存,那么可以持续采样,然后进行批处理。
本发明实施例中,由于此Goertzel基本算法产生的信号在时域是模拟的,因此对它的检测需要先采样形成数字信号。从组成此算法的信号的高,低频率组中可以看到其最大频率为1800Hz,此算法所需的1400HZ和1800HZ频率都位于人的可听范围内,在电信应用中普遍采用8kHz的采样率,即每秒8000个采样,根据奈奎斯特采样定律,可以完全满足要求。
下面结合图2,对本发明具体实施例的通话过程中信息的传送方法做具体说明:(假设该通话过程中要传递的信息的第三方的联系方式)
步骤101,甲乙双方建立通话;
步骤102,通话甲方请求通话乙方发送联系人A的联系方式(电话号码和邮箱地址);
步骤103,判断通话乙方是否记得联系人A的联系方式;
步骤104,若通话乙方记得联系人A的联系方式,则直接调用菜单中专用的软键盘,在软键盘按照顺序输入联系方式号码;
步骤105,若通话乙方不记得联系人A的联系方式,则调用乙方通讯录接口,查询到对应的联系人A,然后在菜单中选中“通过ASCII码发送联系方式”,此时联系人A的联系方式将通过内置的ASCII码编码程序模拟键盘输入方式在后台对联系方式进行按键操作;
步骤106,将代表联系方式的字符和数值的ASCII码频率信号转化为语音信号并混音到发送方向放入PCM语音流中发送给通话甲方。
为了更好的实现上述目的,如图3所示,本发明实施例还提供一种通话过程中信息的获取方法,包括:
步骤4,获取包含ASCII码序列的语音信号;
步骤5,解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
具体的,本发明一具体实施例中,所述获取方法还包括:
步骤6,通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
本发明实施例中,当接收方通讯终端正确获取到对端发送的联系方式的握手信号后,开始实时地去检测语音的PCM流中的信息,当语音流中的PCM语音包含的数字或字母信息满足了此算法要求的频率,时间和幅值,则提取其中的号码信息如0-9,*,#,.,@,A-Z,a-z等。并且产生中断,同时给用户进行完整提示,用户可以在通话中进行实时地保存联系人电话号码和手机号码。这种中断的方式比轮训方式大大地节约了系统资源和功耗。这些特性不仅简化了ASCII码信号的解码同时也降低了ASCII码的误检的概率。这样便使得接收端可以检测出按键的信息并可以摈弃语音和噪声的谐波分量。因此语音和噪声被误检为ASCII码信号的概率就大大减小。
本发明的上述实施例中,如图4所示,步骤5包括:
步骤51,分别获取所述语音信号的行频和列频;
步骤52,根据所述语音信号的行频和列频,获取对应的二进制码序列;
步骤53,分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
步骤54,根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
步骤55,将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
承续上例,本发明具体实施例中,在成功建立通话后,在DSP中实时地检测对端是否发送ASCII码信号,当甲方成功收到包含ASCII码音频序列的语音,则马上调用内置的ASCII码解析程序对语音进行解析。内置的ASCII码解析程序寻找信号的行频和列频,只有符合算法的频率和幅值都满 足了要求,才是正确的ASCII码信号,这样有效过滤了一些语音中的谐波干扰。最终解析的结果是连续的0-9的数值或A-Z,a-z,*,@,#等其他字符,然后在通话界面进行提示给甲方,并调用通讯录接口进行提示保存或放弃。
下面结合图5对本发明实施例的通话过程中信息的获取方法作具体说明:
步骤201,通话甲方收到包含ASCII码序列的语音数据;
步骤202,调用通话甲方的DSP内部解码程序对ASCII码数据进行解析;
步骤203,将解析完的完整的电话号码或邮箱地址显示通话甲方的通话界面上;
步骤204,调用通话甲方的通讯录接口进行联系方式保存提示;
步骤205,通话甲方成功保存解析完的联系人A的联系方式。
具体的,假如传递电话号码13314528888,那么对应的ASCII码的十进制为49,51,51,49,52,53,50,56,56,56,56,则二进制为:110001,110011,110011,110001,110100,110101,110010,111000,111000,111000,111000。
假设传递的邮箱地址为wgh@zte.com.cn,那么对应的ASCII码的十进制为119,103,104,64,122,116,101,46,99,111,109,46,99,110,则二进制为1110111(w),1100111(g),1101000(h),1000000(@),1111010(z),1110100(t),1100101(e),101110(.),1100011(c),1101111(o),1101101(m),101110(.),1100011(c),1101110(n)。
在发送端,ASCII码发送模块将按照代表联系方式的二进制序列,按照算法要求的频率和时间要求产生音频信号,并混音到上行的通话语音流中。在接收端,在收到鉴权信号后,开启ASCII码解析模块的检测算法,对下 行的语音流进行实时检测,首先解析出发送端发送的频率和时间要求满足那一个ASCII码,根据ASCII码查询到对应哪个字符和数字以及等特殊字符,然后根据组合情况决定最终接收到的联系方式,如电话号码和邮箱地址。如果同时传递电话号码和邮箱地址,按照算法会有规定的顺序去发送,两个内容中间会预置一段空白的内容作为分割线,以便正确区分电话号码和邮箱地址。
为了更好的实现上述目的,如图6所示,本发明实施例还提供一种通话过程中信息的传送装置,包括:
获取模块10,配置为获取第通话过程中要发送的信息对应的ASCII码的二进制序列;
转换模块20,配置为根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
发送模块30,配置为将所述语音信号混音到通话语音流中发送给接收方通讯终端。
具体的,本发明一具体实施例中,所述获取模块10包括:
第一获取子模块,配置为通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
第二获取子模块,配置为根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
第三获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
具体的,本发明一具体实施例中,所述获取模块10还包括:
第四获取子模块,配置为调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
第五获取子模块,配置为响应所述通话过程中要发送的信息的传送操 作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
第六获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
具体的,本发明一具体实施例中,所述转换模块20包括:
预设模块,配置为分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
生成模块,配置为利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在预设位置进行音调检测,生成包含ASCII码序列的语音信号。
本发明实施例的传送方法中,通过将通话过程中要发送的信息转化成包含ASCII码的二进制序列的语音信号,并混到通话语音流中发送,实现了无需挂断电话,实时传送信息。需要说明的是,本发明实施例提供的传送装置是应用上述传送方法的装置,则上述传送方法的所有实施例均适用于该装置,且均能达到相同或相似的有益效果。
为了更好的实现上述目的,如图7所示,本发明实施例还提供一种通话过程中信息的获取装置,包括:
语音获取模块40,配置为获取包含ASCII码序列的语音信号;
解析模块50,配置为解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
本发明一具体实施例中,所述获取装置还包括:
保存模块60,配置为通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
具体的,本发明一具体实施例中,所述解析模块50包括:
第一解析子模块,配置为分别获取所述语音信号的行频和列频;
第二解析子模块,配置为根据所述语音信号的行频和列频,获取对应的二进制码序列;
第三解析子模块,配置为分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
第四解析子模块,配置为根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
第五解析子模块,配置为将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
本发明实施例提供的获取方法中,通过接受方解析上述语音信号,并显示该通话过程中要发送的信息,实现了传送信息的便捷性,极大的增强了用户体验;同时采用ASCII码传送的方法具有安全性高及加密鉴权性等特性,具有较好的应用前景。
需要说明的是,本发明实施例提供的获取装置是应用上述获取方法的装置,则上述获取方法的所有实施例均适用于该获取装置,且均能达到相同或相似的有益效果。
为了更好的实现上述目的,本发明实施例还提供一种终端,包括如上所述的通话过程中信息的传送装置和如上所述的通话过程中信息的获取装置。
本发明具体实施例中的传送装置和获取装置需要成对使用才可以完成发送和解析的完整功能,单独一方无法完成整体功能。本专利发明具有安全性高,有加密鉴权等特性,在未来的智能终端的应用中会越来越广泛。
需要说明的是,本发明实施例提供的终端是包括上述传送装置和上述获取装置的终端,则上述传送装置和获取装置的所有实施例及其有益效果均适用于该终端。
为了更好的实现上述目的,本发明实施例还提供一种计算机存储介质,其中存储有计算机可执行指令,所述计算机可执行指令用于执行上述的方法。
上述各模块可以由电子设备中的中央处理器(Central Processing Unit,CPU)、数字信号处理器(Digital Signal Processor,DSP)或可编程逻辑阵列(Field-Programmable Gate Array,FPGA)实现。
本领域内的技术人员应明白,本发明的实施例可提供为方法、系统、或计算机程序产品。因此,本发明可采用硬件实施例、软件实施例、或结合软件和硬件方面的实施例的形式。而且,本发明可采用在一个或多个其中包含有计算机可用程序代码的计算机可用存储介质(包括但不限于磁盘存储器和光学存储器等)上实施的计算机程序产品的形式。
本发明是参照根据本发明实施例的方法、设备(系统)、和计算机程序产品的流程图和/或方框图来描述的。应理解可由计算机程序指令实现流程图和/或方框图中的每一流程和/或方框、以及流程图和/或方框图中的流程和/或方框的结合。可提供这些计算机程序指令到通用计算机、专用计算机、嵌入式处理机或其他可编程数据处理设备的处理器以产生一个机器,使得通过计算机或其他可编程数据处理设备的处理器执行的指令产生用于实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能的装置。
这些计算机程序指令也可存储在能引导计算机或其他可编程数据处理设备以特定方式工作的计算机可读存储器中,使得存储在该计算机可读存储器中的指令产生包括指令装置的制造品,该指令装置实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能。
这些计算机程序指令也可装载到计算机或其他可编程数据处理设备上,使得在计算机或其他可编程设备上执行一系列操作步骤以产生计算机 实现的处理,从而在计算机或其他可编程设备上执行的指令提供用于实现在流程图一个流程或多个流程和/或方框图一个方框或多个方框中指定的功能的步骤。

Claims (16)

  1. 一种通话过程中信息的传送方法,其中,包括:
    获取通话过程中要发送的信息对应的ASCII码的二进制序列;
    根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
    将所述语音信号混音到通话语音流中发送给接收方通讯终端。
  2. 根据权利要求1所述的通话过程中信息的传送方法,其中,所述获取通话过程中要发送的信息对应的ASCII码的二进制序列,包括:
    通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
    根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
    将所述十进制码转化为对应的ASCII码的二进制序列。
  3. 根据权利要求1所述的通话过程中信息的传送方法,其中,所述获取通话过程中要发送的信息对应的ASCII码的二进制序列的步骤还包括:
    调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
    响应所述通话过程中要发送的信息的传送操作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
    将所述十进制码转化为对应的ASCII码的二进制序列。
  4. 根据权利要求1所述的通话过程中信息的传送方法,其中,根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号,包括:
    分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
    利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在预设位置进行音调检测,生成包含ASCII码序列的语音信号。
  5. 一种通话过程中信息的获取方法,其中,包括:
    获取包含ASCII码序列的语音信号;
    解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
  6. 根据权利要求5所述的通话过程中信息的获取方法,其中,所述获取方法还包括:
    通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
  7. 根据权利要求5所述的通话过程中信息的获取方法,其中,解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示,包括:
    分别获取所述语音信号的行频和列频;
    根据所述语音信号的行频和列频,获取对应的二进制码序列;
    分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
    根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
    将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
  8. 一种通话过程中信息的传送装置,其中,包括:
    获取模块,配置为获取第通话过程中要发送的信息对应的ASCII码的二进制序列;
    转换模块,配置为根据预设的二进制数的频率和时间,将所述ASCII码的二进制序列转化成语音信号;
    发送模块,配置为将所述语音信号混音到通话语音流中发送给接收方通讯终端。
  9. 根据权利要求8所述的通话过程中信息的传送装置,其中,所述获取模块包括:
    第一获取子模块,配置为通过发送方通讯终端的软键盘依次键入通话过程中要发送的信息;
    第二获取子模块,配置为根据ASCII码表格,获取所述通话过程中要发送的信息对应的ASCII码的十进制码;
    第三获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
  10. 根据权利要求8所述的通话过程中信息的传送装置,其中,所述获取模块还包括:
    第四获取子模块,配置为调用发送方通讯终端的通讯录接口,获取通话过程中要发送的信息;
    第五获取子模块,配置为响应所述通话过程中要发送的信息的传送操作,根据ASCII表格,将所述通话过程中要发送的信息转化为对应的ASCII码的十进制码;
    第六获取子模块,配置为将所述十进制码转化为对应的ASCII码的二进制序列。
  11. 根据权利要求8所述的通话过程中信息的传送装置,其中,所述转换模块包括:
    预设模块,配置为分别预设二进制数1和二进制数0的频率,以及该二进制数1和二进制数0的预设频率的持续时间;
    生成模块,配置为利用Goertzel基本算法对所述ASCII码的二进制序列进行处理,并在预设位置进行音调检测,生成包含ASCII码序列的语音信号。
  12. 一种通话过程中信息的获取装置,其中,包括:
    语音获取模块,配置为获取包含ASCII码序列的语音信号;
    解析模块,配置为解析所述语音信号,得到通话过程中要发送的信息并在接收方通讯终端的通话界面上显示。
  13. 根据权利要求12所述的通话过程中信息的获取装置,其中,所述获取装置还包括:
    保存模块,配置为通过调用所述接收方通讯终端的通讯录接口保存所述通话过程中要发送的信息。
  14. 根据权利要求12所述的通话过程中信息的获取装置,其中,所述解析模块包括:
    第一解析子模块,配置为分别获取所述语音信号的行频和列频;
    第二解析子模块,配置为根据所述语音信号的行频和列频,获取对应的二进制码序列;
    第三解析子模块,配置为分别将所述行频和列频的频率和幅值与预设范围进行比较,当所述频率和幅值均在所述预设范围内时,确定该二进制码序列为ASCII码序列;
    第四解析子模块,配置为根据所述ASCII码序列,得到对应的通话过程中要发送的信息;
    第五解析子模块,配置为将所述通话过程中要发送的信息在接收方通讯终端的通话界面上显示。
  15. 一种终端,其中,包括如权利要求8至11任一项所述的通话过程中信息的传送装置和如权利要求12至14任一项所述的通话过程中信息的获取装置。
  16. 一种计算机存储介质,其中存储有计算机可执行指令,所述计算机可执行指令用于执行所述权利要求1至4、权利要求5至7任一项所述的方法。
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