TWI569592B - Information transmission system for improving the correctness of data and its data processing - Google Patents

Description

Audio transmission system for improving data correctness and data processing method thereof

The present invention relates to an audio transmission system, and more particularly to an audio transmission system and a data processing method thereof for improving data correctness.

With the rapid development of technology, the transmission technology of transmitting and receiving information through mobile devices' 3G or Wi-Fi is quite common. In recent years, there are more technologies that use sound to transmit information (such as ultrasonic communication technology), such as transmitting URL information. As long as the user holds the mobile phone near the TV, they can play related information, or they can receive relevant information when they see the concert video, or purchase information. The retail store can also produce some electronic coupons, etc. . In the current ultrasonic communication technology, it usually adopts the frequency modulation method, that is, the frequency of the bit is represented by "0" or "1" of the bit value, but the frequency modulation mode has the disadvantage of slow transmission rate, when the frequency changes. When it is too dense, the transmission of sound waves in the air will be affected by the uneven propagation rate, and the sound waves of different frequencies will arrive at the device for receiving at the same time, so that the change of the acoustic frequency cannot be correctly identified, so it must be inefficient. Change the frequency at intervals.

For example, in the prior art, there is a system for transmitting data and control commands by using sound. Referring to FIG. 8 , a broadcast device 91 and a sound pickup device 92 are provided. The user operates on the broadcast device 91 and inputs a string or Commands and the like, and converted into a playable sound file to transmit sound waves to the sound pickup device 92 by air, and the sound pickup device 92 receives the sound waves and then processes the sound waves to obtain strings or commands in the sound waves, and Automatically execute the corresponding action according to the content of the recognized string or instruction, let The user can transmit the sound using only the sounding device 91, and can control the corresponding sound pickup device 92 and obtain the required service. As for the prior art broadcast device 91, the user input string is converted into sound waves for transmission, and is received by the corresponding sound receiving device 92, as shown in FIG. 9 and FIG. 10, wherein the sounding device 91, the The radio device 92 performs the following steps: the broadcast device 91 accepts a string input by the user, converts the string input by the user into binary data, and modulates the binary data through a phase shift (Phase Shift Keying, PSK) converts the binary data into a sound file; the broadcast device 91 determines whether to compress the sound file according to the preset mode of the system, and if so, performs destructive/non-distortion compression and plays the compressed file. If not, the sound file is directly played; when the sound receiving device 92 receives a set of sound signals, the set of sound signals is demodulated and converted into a string, and then determined according to the content of the string. Follow-up actions taken.

According to the principle of the phase shift modulation technique, the change of the phase of the acoustic wave mainly represents the "0" or "1" of the bit value, so that the frequency is not changed when transmitting, and the interval time for transmitting different bits can be greatly reduced. In order to improve the problem that the prior art must change the frequency more inefficiently for a period of time, the initial design object of the phase shift modulation technique is mainly electromagnetic wave transmission, and the propagation speed of the electromagnetic wave is the speed of light, so in the air. The phase shift generated after transmitting a distance is extremely small. In contrast, the sound wave propagates at the speed of sound in the air. Compared with the speed of light, the speed of sound is about 880,000 times slower, and the sound wave is transmitted in the medium due to the difference in density of the medium. The change of the propagation speed, therefore, after a certain distance is transmitted in the air, the phase offset will increase. If the radio device 92 directly interprets, a large number of consecutive error bits will be read, so the phase offset is adjusted. Although the variable technique does not have the problem of changing the frequency, the correctness of the data is reduced.

It can be known from the above prior art that in order to improve the accuracy of data transmission, the general transmission system generally improves the communication quality of the data transmission channel and reduces the probability of transmitting error information, but the transmission of sound waves in the air belongs to the signal-to-noise ratio ( Signal to Noise Ratio, SNR) low instability channel, through the current state of the art sound The mode of wave transmission needs to be transmitted before the data is modulated by the sounding device 91, and the transmitted sound wave will change the phase of the audio abnormally due to the difference of the medium or the distance based on the physical characteristics, and generate a large amount of Continuity of the bit data, and can not be corrected, so that after receiving the sound wave, the radio device 92 can not judge the correct information and can only choose to discard and retransmit the data; in addition, in order to enhance the sound wave The efficiency of transmission, when using the phase shift modulation technique, the sound wave often generates a large amount of continuous error bit data in the air transmission system, which can not only obtain the correct data, but also can not be corrected, so that the system efficiency is not good.

Therefore, with the above prior art, it is indeed necessary to propose a more ideal solution.

In view of the above-mentioned deficiencies of the prior art, the main object of the present invention is to provide an audio transmission system for improving data correctness and a data processing method thereof, which perform sound through a device for transmitting sound and a device for receiving sound. Transmission, after the information to be transmitted is re-arranged and broken up in a specific way, in order to avoid the sudden error in the transmission process, the error bit after receiving is too concentrated, so that it cannot be corrected. It can improve the correctness of obtaining hidden data, and solve the problem that continuous digital data cannot be corrected due to continuous errors during transmission.

The main technical means for achieving the above purpose is to enable the data processing method of the audio transmission system for improving the correctness of the data, mainly by transmitting a sound wave from a first device to a matching second device, and by the first device Performing the following steps: accepting a message and converting the information into a digital data; arranging the digital data into a matrix; performing an error correction coding algorithm such that the continuous data in the matrix is re-encoded to generate a word Meta-sequence data; adding a header information to the character sequence data to form a meta-arrangement data, And modulating the bit arrangement data into a set of sound waves; playing the set of sound waves for reception by the second device.

The above steps are performed by the first device sound wave and received by the second device. When the first device receives the information, and converts the information into the digital data, the digital data is arranged into the matrix. A device performs the error correction encoding algorithm such that successive data in the matrix is re-encoded to generate the character sequence data, and the first device adds the header information to the header information to form the The bit arrangement data, and the bit arrangement data is modulated into a set of sound waves for reception by the second device; when the second device receives the set of sound waves, it only needs to filter, demodulate and obtain The bit arrangement data, the character sequence data is identified by the header information in the bit arrangement data, and an error correction corresponding to the decoding algorithm is performed on the character sequence data to obtain correct information, thereby improving Obtain the correctness of the hidden data, and solve the problem that the continuous digital data cannot be corrected due to continuous errors during the transmission process.

Another main technical means for achieving the above purpose is that the audio transmission system for improving the correctness of the data includes: a first device comprising a first processor, an audio output unit and an encoder, the first The processor is respectively connected to the audio output unit and the encoder, and outputs audio through the audio output unit; a second device is matched with the first device, and the second device includes a second processor and a second device. An audio receiving unit and a decoder, wherein the second processor is respectively connected to the audio receiving unit and the decoder, wherein the audio receiving unit is configured to receive audio and transmit the audio to the second processor, where the second processor Performing audio processing and demodulation; wherein, when the first processor of the first device reads a message, the information is converted into a digital data, and the digital data is arranged into a matrix, and then the encoder Performing an error correction coding algorithm such that successive data in the matrix is re-encoded to generate a sequence of character data, and the first processor adds the sequence of the character A header information to form one yuan sort the data, and the bit financing arrangement The material is modulated into a set of sound waves, and the set of sound waves is played by the audio output unit for receiving by the second device.

According to the above configuration, the audio transmission system of the present invention improves the correctness of the data through the first processor of the first device, and the encoder transmits the processed audio through the audio output unit to play a group of sound waves containing information. The audio receiving unit of the second device receives the set of sound waves, and then converts the set of sound waves into audio signals for transmission to the second processor, and performs subsequent corresponding audio processing or data identification via the second processor and the decoder. The manner in which the first device processes the audio is to read the information through the first processor, convert the information into a digital data, and arrange the digital data into a matrix format, and then perform the error correction by the encoder. Encoding algorithm, such that the continuous data in the matrix is re-encoded to generate the character sequence data, and the first processor adds the header information to the header information to form the bit alignment data, and The bit arrangement data is modulated into a set of sound waves, and the set of sound waves is played by the audio output unit for receiving by the second device, and the audio connection of the second device is After receiving the group of sound waves, the unit filters, demodulates and demodulates the data through the second processor, and obtains the bit sequence data through the header information in the bit arrangement data. And performing, by the decoder, performing an error correction on the character sequence data to obtain a correct information, thereby improving the correctness of obtaining the hidden data, and solving the continuous error of the continuous digital data in the transmission process. The purpose of causing problems that cannot be corrected.

10‧‧‧ first device

11‧‧‧First processor

12‧‧‧Audio output unit

13‧‧‧Encoder

20‧‧‧second device

21‧‧‧second processor

22‧‧‧Optical receiving unit

23‧‧‧Decoder

91‧‧‧Broadcasting equipment

92‧‧‧ Radio equipment

1 is a schematic diagram of an audio transmission system in accordance with a preferred embodiment of the present invention.

2 is a schematic diagram showing the structure of an information coding arrangement according to a preferred embodiment of the present invention.

3 is a flow chart of a first device data processing method according to a preferred embodiment of the present invention.

4 is a flow chart of a first device error correction coding algorithm in accordance with a preferred embodiment of the present invention.

FIG. 5 is a flow chart of a second device data processing method according to a preferred embodiment of the present invention.

6 is a flow chart of demodulation and information conversion of a second device according to a preferred embodiment of the present invention.

FIG. 7 is a flow chart of a second device error correction corresponding decoding algorithm according to a preferred embodiment of the present invention.

Figure 8 is a block diagram of a known audio transmission system.

Figure 9 is a flow chart showing the playback sound of a known broadcast device.

Figure 10 is a flow chart of the received sound of a known radio device.

For a preferred embodiment of the audio transmission system for improving the audio recognition rate of the present invention, refer to FIG. 1, which is mainly for transmitting a set of a first device 10 to a second device 20 in a medium (such as air). Sound waves, and the first device 10 is matched with the second device 20, wherein the sound wave (sound) is a continuous signal in the air, which may be called an audio signal (Audio Signal), and is also referred to as a person. The signal of the various sounds heard by the ear. In this embodiment, the first device 10 and the second device 20 are respectively a mobile device, a smart device, or an electronic device such as a computer device.

The first device 10 includes a first processor 11, an audio output unit 12, an encoder 13 and an input unit (not shown). The first processor 11 and the audio output unit 12 respectively. The encoder 13 is connected to the input unit, and the first processor 11 can input/preset information or input information through the input unit, and the first processor 11 of the first device 10 reads a message, and Converting the information into information, converting the information into a digital data, adding a check code to the digital data and arranging it into a matrix, and then performing an error correction coding algorithm by the encoder 13 to make the matrix The continuous data is re-encoded to generate a character sequence data, and the first processor 11 adds a header information to the character sequence data to form a bit-array data, when the first processor 11 After reading the bit arrangement data, it is converted into a set of sound waves, and the set of sound waves is played through the audio output unit 12 and transmitted in the air to output a sound wave containing information for the second device 20 to be connected. Received.

The second device 20 includes a second processor 21, an audio receiving unit 22, and a decoder 23. The second processor 21 is respectively connected to the audio receiving unit 22 and the decoder 23. The audio receiving unit is connected to the audio receiving unit. 22 is configured to receive the sound played by the first device 10, and transmit the received set of sound waves to the second processor 21, and the second processor 21 performs audio processing and demodulation; After receiving the sound wave containing the information, the second device 20 filters and demodulates the sound wave of the group through the second processor 21, and obtains the bit arrangement data containing the header information, and arranges the data according to the bit. The header information is used to identify the character sequence data, and the decoder 23 performs an error correction corresponding to the decoding algorithm on the character sequence data, and the error correction corresponding to the decoding algorithm matches the error correction coding algorithm. Therefore, it is possible to accurately obtain the correct information. Since the sound waves are susceptible to various influences during the transmission process, it is prone to a large number of consecutive data errors that cause uncorrectable problems. The encoder 13 of the first device 10 executes the error correction coding algorithm such that the continuous data re-encoding arrangement breaks up the data arrangement in a specific mathematical manner, avoiding the error bit after reception caused by sudden continuous errors in the transmission process. Too much concentration and cannot be corrected, and the decoder 23 of the second device 20 performs matching error correction corresponding to the decoding algorithm, which not only can obtain correct information, but also can increase the accuracy and performance of the system.

To illustrate the processing manner of the information structure of the information encoding arrangement of the first device 10 in the preferred embodiment of the present invention, refer to FIG. 2, in which the information is read by the first processor 11 of the first device 10. In this embodiment, the information may be an input string. When the information conversion is performed, the input string is first converted into the digital data S corresponding to the ASCII code, and the digital data S includes a set of binary bit values. Adding the above check code to the digital data S and rotating it to be arranged as a matrix S0 ("S0:[0][1][2][3]..."), and then using the encoder 13 to the matrix S0 Performing the above error correction coding algorithm, and performing a transposition operation on the matrix format in the rotated matrix S0, that is, the horizontal axis and the vertical axis of the matrix format are exchanged to form a transposed matrix S1. ("S1:[0][1][2][3]..."), and then the matrix S0 and the data of the transposed matrix S1 are continuously re-interleaved and encoded to generate a sequence of characters. Data S2 ("S0[0], S1[0], S0[1], S1[1], S0[2], S1[2],..."), and finally add the character sequence data S2 The header information constitutes a one-bit arrangement data S3, and the bit arrangement data is modulated into a set of sound waves for playback to the second device 20.

When the second processor 21 of the second device 20 receives the set of sound waves, performs filtering, demodulation, and obtains the bit array data S3 containing the header information, and arranges the headers in the data S3 according to the bits. The information reads the content of the character sequence data S3, and the corresponding error correction corresponding decoding algorithm is executed by the decoder 23, and the content of the read character sequence data S3 is disassembled and separated, and the character is separated. The matrix S0 and the transposed matrix S1 in the content of the sequence data S3 are restored to obtain the data of the error bit value, and all possible paths are obtained by using a trellis diagram, and then a minimum Hamming is taken. The path of the Hamming distance is the correct information content for maximum probability. In this embodiment, the second processor 21 filters and demodulates the set of sound waves and obtains the character sequence data having the temporal state characteristic, wherein the second processor 21 further transmits a band pass filter, A Fast Fourier Transform (FFT) is used to separately filter and demodulate the set of sound waves.

In summary, the main application mode of the present invention is to first convert a string of characters into a binary bit value by the first device 10, and then use a forward error correction (FEC) for the binary bit value. The technology encodes and performs modulation after the encoding. After the modulation is completed, the modulated audio signal is transmitted to the second device 20; when the second device 20 receives the audio signal, the audio signal is further processed. Demodulation is changed and decoded after demodulation, and the decoded data is converted into a string. When the sound wave is transmitted through the forward error correction technique, a large number of erroneous bit data can be corrected to be correct. It must be noted that the forward error correction technique is a generic term for a technique in which redundant bits are added in a specific format in the data so that the data in error during transmission can be corrected. Including Reed-Solomon Code, Convolutional Code, Turbo Code, etc.

In addition, since the general communication system is designed to improve the accuracy of data transmission, it is usually necessary to improve the communication quality of the data transmission channel to reduce the probability of transmitting error information. However, since the sound wave is transmitted in the air, it belongs to the signal-to-noise ratio (Signal). To Noise Ratio (SNR) is a low unstable channel. Therefore, in this embodiment, the error bit caused by the transmission process is corrected by using an error correction coding technique, and the best error correction coding algorithm and matching The error correction corresponds to the decoding algorithm, which is a convolutional code (also known as Convolutional Code) and a Viterbi algorithm decoding.

Further, the error correction coding algorithm and the matching error correction corresponding to the decoding algorithm may further be a lattice coding and a Viterbi algorithm decoding, an acceleration code (Turbo Code) coding and a Witt. Ratio algorithm decoding, a linear code encoding, and a Viterbi algorithm decoding.

According to the description of the preferred embodiment of the present invention, the data processing method of the audio transmission system for improving the correctness of the data can be summarized, mainly by the first device 10 transmitting sound waves to the matched second device 20, such as As shown in FIG. 2 and FIG. 3, the first device 10 performs the following steps: receiving a message (S31). In this embodiment, the information can be a string and converting the information into a digital data S (S32). The digital data S includes a set of binary bit values; the digital data is arranged into a matrix S0 (S33). In this embodiment, a check code may be added to the digital data and arranged as the matrix S0 (S33). Performing an error correction encoding algorithm (S34) such that the continuous data in the matrix S0 is re-encoded to generate a character sequence data S2; the character sequence data S2 is added with a header information to form a The bit arrangement data S3, and the bit arrangement data S3 is modulated into a set of sound waves (S35); the set of sound waves is played for the second device 20 to receive, or the set of sound waves is stored (S36); In the example, the sound wave of the group is further compressed; whether it is required to be produced Give birth to a set of sound waves (S37), and if so, go back to the above Subject to a message (S31).

By the above steps, the first device 10 plays a sound to the second device 20, and when the first device 10 receives the information, converts it into a digital data S, and when the error correction coding algorithm is executed. The continuous data in the matrix S0 is re-encoded to generate the character sequence data S2. In the present embodiment, when the method is executed to "execute an error correction coding algorithm (S34), the continuous data in the matrix S0 is made. When the steps are re-encoded to generate a character sequence data S2", as shown in FIG. 2 and FIG. 4, the method further includes the following steps: the digital data S containing the check code is rotated in a fixed direction to form the matrix S0. (S341); converting the matrix S0 to generate the transposed matrix S1 (S342); performing continuous re-interlacing coding on the matrix S0 and the data of the transposed matrix S1 into the character sequence data S2 In this embodiment, the character sequence data S2 is a one-dimensional array S2 (S343); and the foregoing "adding a header information to the character sequence data S2 to form a one-dimensional array data S3" The bit arrangement data S3 tone It is a set of sound waves (S35); "step and" the set of acoustic player for the second receiving means 20, or (S36) "step of storing the set of sound waves.

By adding the header information to the character sequence data S2 in the first step 10 to form the bit arrangement data S3, and converting the bit arrangement data S3 into a set of sound waves for the first The second device receives 20, in the embodiment, when the first device 10 plays the set of sound waves and is received by the second device 20, as shown in FIG. 5, and further, the second device 20 performs the following steps: accepting a set of sound waves (S41); the set of sound waves is filtered and converted (S42). In this embodiment, the second device 20 is further transmitted through a band pass filter and a Fast Fourier Transform (FFT). Filtering and demodulating the set of sound waves respectively; demodulating the converted sound feature into a digital data (S43); reading the character sequence data S2 according to the header information in the bit arrangement data S3 And performing an error correction on the character sequence data S2 corresponding to the decoding algorithm to obtain correct information (S44); converting the information into a string and returning the information The near/far end (S45) determines whether or not to continue receiving the next set of sound waves (S46), and if so, returns to the above-mentioned "accepting a set of sound waves (S41)" step.

Furthermore, when the above step is performed to "read the content of the character sequence data S2 according to the header information in the bit arrangement data S3, and perform an error correction corresponding decoding algorithm on the character sequence data S2 to obtain In the correct information (S44) step, as shown in FIG. 6, and the second device 20 further performs the following steps: reading the converted sound characteristics of the set of sound waves (S441); demodulating the sound features Changing (S442) to generate the digital data; determining, according to the digital data, whether one of the header information is read (S443); if so, reading the character sequence data S2 (S444); The sequence data S2 performs the error correction corresponding to the decoding algorithm to obtain correct information (S445); it is again determined by a check code whether the digital data S is correct (S446); if yes, the foregoing "converting the information into one And returning the string to the near/far end (S45) step; when performing the foregoing step of "determining whether one of the header information is read according to the digital data (S443)", or when performing the foregoing "with a check code" Again judge the calculation of the digit If the result of the data S is correct (S446), if the result of the determination is no, the range in which the set of sound waves is read is changed (S447), and the process returns to the step of "reading the converted sound characteristics of the set of sound waves (S441)". .

Further, in the embodiment, when the above step is performed to the step of "execution of the error correction corresponding decoding algorithm on the character sequence data S2 to obtain correct information (S445)", as shown in FIG. 2 and FIG. The method further includes the steps of: taking out the corresponding matrix S0 and the transposed matrix S1 (S4451) in the data content of the character sequence data S2; and performing the reduction arrangement on the matrix S0 and the transposed matrix S1 (S4452); The sequence comparison restores the data contents of the aligned matrix S0 and the transposed matrix S1 to obtain the data of the error bit value (S4453); uses the Viterbi algorithm to decode and obtain the correct digital data S content. (S4454); Continuing to execute the above-mentioned "determining whether the digital data S is correct by a check code (S446)" step.

The invention corrects the corresponding decoding algorithm through the error correction coding algorithm and the corresponding error correction, calculates the shortest path according to the data of the error bit value, and obtains the correct information content of the maximum probability, and is based on the current prior art general mobile phone. The computing power of the chip is not yet able to perform too complicated operations. Therefore, the technique of error correction is previously encoded, and then decoded by the corresponding Viterbi algorithm, so that a large number of erroneous bit data can be corrected when the sound wave is transmitted. Therefore, when the phase shift modulation technique is used in the prior art, the sound wave often generates a large amount of continuous error bit data in the air transmission system, which causes an uncorrectable problem, and can improve the correctness of the hidden data obtained by the system. effectiveness.

10‧‧‧ first device

11‧‧‧First processor

12‧‧‧Audio output unit

13‧‧‧Encoder

20‧‧‧second device

21‧‧‧second processor

22‧‧‧Optical receiving unit

23‧‧‧Decoder

Claims (11)

A data processing method for an audio transmission system for improving data correctness mainly comprises: transmitting, by a first device, a sound wave to a matching second device, and performing, by the first device, the following steps: receiving a message and the information Converting to a digital data; rotating the digital data in a fixed direction to be arranged into a matrix; performing an error correction coding algorithm such that successive data in the matrix are re-encoded to generate a character sequence data; Converting a matrix to generate a transposed matrix; interleaving the data of the matrix and the transposed matrix into the sequence of the character sequence; adding a header information to the sequence of the character sequence to form a bit-array data, And modulating the bit arrangement data into a set of sound waves; playing the set of sound waves for reception by the second device. For the data processing method of the audio transmission system for improving the correctness of the data described in claim 1, when the foregoing steps are performed to arrange the digital data into the matrix step, the following steps are further performed: adding a check code to the digital data. And arranged as the matrix. The data processing method for an audio transmission system for improving the correctness of data according to claim 1, the method further comprising the steps of: compressing the group of sound waves, and determining whether a next set of sound waves is to be generated, and if so, returning to the foregoing Accept an information step. The data processing method of the audio transmission system for improving the correctness of the data, as described in claim 1, wherein the information may be a string, and the information is converted into the digital data, and the digital data includes a set of binary bit values. The data processing method of the audio transmission system for improving data correctness as described in claim 1, wherein the character sequence data is a one-dimensional array. The data processing method of the audio transmission system for improving the correctness of the data according to any one of claims 1 to 5, wherein in the above method, performing the step of playing the set of sound waves to the first device and receiving the second device And further performing, by the second device, the following steps: accepting the set of sound waves; converting the set of sound waves; demodulating the converted sound features into the digital data; and reading the character sequence data according to the header information Content, and perform an error correction corresponding to the decoding algorithm to get the right information. The data processing method of the audio transmission system for improving the correctness of the data as described in claim 6, when the above steps are performed to perform the conversion step of the set of sound waves, and further comprising the steps of: filtering and converting the set of sound waves. The data processing method of the audio transmission system for improving the correctness of the data as described in claim 6, the method further comprising the steps of: converting the information into a string and transmitting back to the near/far end; determining whether to continue receiving the next set of sound waves. If yes, go back to the previous step of accepting a set of sound waves. The data processing method of the audio transmission system for improving the correctness of the data as described in claim 8, when the step is performed to read the content of the character sequence according to the header information, and performing the error correction corresponding to the decoding algorithm, Obtaining the correct information step, and further performing the following steps: reading the sound characteristics of the set of sound waves; demodulating the sound feature to generate the digital data; determining whether the header information is read according to the digital data If yes, the content of the character sequence data is read; the error correction is performed corresponding to the decoding algorithm to obtain correct information; and it is determined again whether the digital data is calculated correctly; If yes, performing the foregoing conversion of the information into a string and returning to the near/far end step; performing the foregoing step of determining whether the header information is read according to the digital data, or calculating the digit when performing the foregoing re-determination If the data is correct, if the result of the determination is no, the range in which the set of sound waves is read is changed, and the sound characteristic step of reading the set of sound waves is returned. The data processing method of the audio transmission system for improving the correctness of the data as described in claim 9, when the above steps are performed to perform the error correction corresponding to the decoding algorithm to obtain the correct information step, the method further comprises the following steps: The matrix and the transposed matrix in the data content of the character sequence data; the aligned matrix and the transposed matrix content are sequentially compared to obtain the error bit data; the Viterbi algorithm is used to decode to obtain the correct digit Data content; continue to perform the foregoing steps to determine whether the digital data is correct. An audio transmission system for improving data accuracy, comprising: a first device, comprising: a first processor, an audio output unit and an encoder, wherein the first processor and the audio output unit and the encoder respectively Connecting, and outputting audio through the audio output unit; a second device is matched with the first device, the second device includes a second processor, an audio receiving unit, and a decoder, the second processing The device is connected to the audio receiving unit and the decoder, and the audio receiving unit is configured to receive audio and transmit the audio to the second processor, where the second processor performs audio processing and demodulation; The first device performs the data processing method of the audio transmission system for improving the correctness of the data according to any one of claims 1 to 5; and the second device performs the audio transmission for improving the correctness of the data as described in claim 6 System data processing method.