CN112291020A - Full-duplex underwater sound digital voice communication system and method thereof - Google Patents
Full-duplex underwater sound digital voice communication system and method thereof Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B13/00—Transmission systems characterised by the medium used for transmission, not provided for in groups H04B3/00 - H04B11/00
- H04B13/02—Transmission systems in which the medium consists of the earth or a large mass of water thereon, e.g. earth telegraphy
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/16—Vocoder architecture
- G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L27/00—Modulated-carrier systems
- H04L27/26—Systems using multi-frequency codes
- H04L27/2601—Multicarrier modulation systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L27/00—Modulated-carrier systems
- H04L27/26—Systems using multi-frequency codes
- H04L27/2601—Multicarrier modulation systems
- H04L27/2626—Arrangements specific to the transmitter only
- H04L27/2627—Modulators
- H04L27/2628—Inverse Fourier transform modulators, e.g. inverse fast Fourier transform [IFFT] or inverse discrete Fourier transform [IDFT] modulators
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L5/00—Arrangements affording multiple use of the transmission path
- H04L5/14—Two-way operation using the same type of signal, i.e. duplex
Abstract
A full-duplex underwater sound digital voice communication system and a method thereof relate to the field of underwater sound communication. The system consists of two parties of communication A and B, which are respectively marked as U1And U2(ii) a The underwater acoustic transducer adopts a round tube type broadband transducer, and the bandwidth of the transducer is divided into B1、B2And B3Equal 3 different sub-bands, of which sub-band B1And B3Having the same bandwidth W1，B2Has a bandwidth of W2；U1And U2The hardware part of the voice coder consists of a voice input and output device, a low-speed vocoder and peripheral circuits thereof, a DSP and a circuit outside the DSPThe device comprises a peripheral circuit, a power amplifier and a matching circuit thereof, a preamplifier, a band-pass filter, a power management module, an underwater acoustic transducer, a hydrophone and the like. The real-time full duplex communication function can be realized. The system has small time delay, is suitable for the fast time-varying underwater acoustic channel environment, and the speech synthesized by the receiving end has high intelligibility, naturalness and definition. The system is portable, stable, low in power consumption and low in price. The system is scalable, easy to debug and high in maintainability.
Description
Technical Field
The present invention relates to the field of underwater acoustic communication, and in particular, to a full duplex underwater acoustic Digital voice communication system and method based on a vocoder and a Digital Signal Processor (DSP).
Background
The underwater voice communication has important application value in civil use and military use. Early underwater voice communication mainly adopts a cable mode, although the communication mode is simple and efficient, and the output voice has high intelligibility, naturalness and definition, the moving range of underwater operation personnel is limited by the length of the cable. The cable is too long, or in some complex marine environments, twisted cables or broken cables may be formed, resulting in the failure of voice communication.
In recent years, with the continuous and deep ocean research and development, underwater wireless voice communication is more and more emphasized by people. Because the attenuation speed of electromagnetic waves and light waves in water is high, the underwater voice communication at a long distance can be carried out only by utilizing sound waves. The underwater acoustic voice communication mainly comprises two modes of single-side band analog modulation and digital modulation, wherein the former mode is AN AN/WQ-2A single-side band voice communication machine in active service of the American navy; the G732MKII type single sideband communicator developed in the UK and used for communication between the submarine and the surface vessel; the latter is a underwater sound digital voice communication system based on mixed code excited linear prediction voice coding and coherent modulation developed by Guo Zhong Source and the like (Guo Zhong Source, Chen rock, Jianing, Guo Jie, Chen 36179, Mofu Source, horsepower. research and implementation of underwater digital voice communication system, acoustical science newspaper, 2008, 33 (5): 409-; an underwater voice digital voice communication system based on voice compression coding and decoding and Orthogonal Frequency Division Multiplexing (OFDM for short) multi-carrier modulation technology developed by sun zong et al (sun zong, george, mawei, malal, yankee, zhongfen, von snowfly, liu rime, patent CN 201310442083.5). Due to the complex and variable characteristics of the marine environment, serious multipath and noise interference exist, the quality of the voice output by the single-sideband analog modulation underwater voice communication system is difficult to guarantee, and the voice is fuzzy under most conditions. Moreover, because of the analog modulation mode, the system has the defects of large size, low power utilization efficiency, easy crosstalk among different users and the like. The underwater sound digital voice communication can overcome the defects of analog voice communication, but the software and hardware involved in the realization of the underwater sound digital voice communication are relatively complex. Document 3 (liu sheng, xu chu mei, xiao sheng yang. a channel adaptive underwater digital voice communication system and method thereof, patent CN201410220208.4) proposes a channel adaptive underwater digital voice communication system and method thereof. The method adaptively selects the OFDM modulation or FH-MFSK modulation mode according to the underwater acoustic channel condition and the ocean noise condition, can ensure the voice quality under a higher signal-to-noise ratio, and can meet the requirements of underwater voice communication under a long distance and a low signal-to-noise ratio.
Although research on underwater acoustic digital voice communication has been advanced, most of the existing systems are in a half-duplex mode, i.e., only in a transmitting or receiving state at a certain time. In order to overcome the defects of the existing underwater sound digital voice communication system, the bandwidth of an underwater sound transducer is divided into 3 different sub-bands, wherein 1 sub-band is used for sending voice data, 1 sub-band is used for receiving the voice data, and 1 sub-band is used for estimating the noise power.
Disclosure of Invention
The invention aims to overcome the defects of the existing underwater sound digital voice communication system and provide a full-duplex underwater sound digital voice communication system and a method thereof which are suitable for a time-varying underwater sound channel.
The full-duplex underwater sound digital voice communication system consists of a communication party A and a communication party B which are respectively marked as U1And U2；U1And U2The positions of the voice signal receiving and transmitting units are completely the same, and the voice signal receiving and transmitting units can independently and autonomously send and receive voice signals, so that the full-duplex voice communication function is realized; the underwater acoustic transducer adopts a round tube type broadband transducer, and the bandwidth of the transducer is divided into B1、B2And B3Equal 3 different sub-bands, of which sub-band B1And B3Having the same bandwidth W1，B2Has a bandwidth of W2(ii) a To communication party U1，B1For the uplink sub-band, for transmitting speech data, B2Is as followsA line subband for receiving voice data; to U2Then, in contrast, B2For the uplink sub-band, for transmitting speech data, B1Is a downlink sub-band for receiving voice data; sub-band B3For estimating the noise power;
U1and U2The hardware part of the system consists of parts such as voice input and output equipment, a low-rate vocoder and peripheral circuits thereof, a DSP and peripheral circuits thereof, a power amplifier and matching circuits thereof, a preamplifier, a band-pass filter, a power management module, an underwater acoustic transducer, a hydrophone and the like; the voice input and output device is used for picking up and playing voice, and the vocoder realizes parameter compression coding and decoding of the voice; the DSP comprises two subtasks besides a main program, wherein the subtask 1 realizes transmitting underwater sound signal processing including channel coding, OFDM modulation and the like, and the subtask 2 realizes receiving underwater sound signal processing including functions of synchronization, Doppler frequency shift estimation and compensation, self-adaptive channel estimation and equalization, channel decoding and the like; each frame signal comprises 1 synchronization head and n OFDM symbols, wherein the 1 st OFDM symbol is transmitted by pilot symbols, and the 2 nd to nth OFDM symbols are transmitted by voice data symbols.
The full-duplex underwater sound digital voice communication method comprises the following steps:
1)U1inputting the voice signal picked up by the microphone into a low-speed vocoder, performing parameter compression coding on the voice signal by the vocoder, and extracting voice coding data according to frames;
2)U1the vocoder transmits the voice coded data obtained in the step 1) to the DSP through an RS232 serial port;
3)U1a subtask 1 of the DSP performs framing, channel coding, symbol mapping, pilot symbol insertion, IFFT conversion and other baseband signals on the voice data obtained in the step 2), and inserts a section of frequency modulation signals with the same length and start-stop frequency before and after each frame signal; each frame signal comprises n OFDM symbols, the 1 st OFDM symbol is all pilot symbols, and the rest n-1 OFDM symbols are all voice data symbols;
4)U1the DSP subtask 1 uses the baseband signal obtained in the step 3)Frequency shift to subband B1The frequency range of (d);
5)U1the DSP subtask 1 outputs the signal obtained in the step 4) to a power amplifier through a D/A converter, and the electric signal after power amplification excites an underwater acoustic transducer to emit sound waves to be transmitted in an underwater acoustic channel;
6)U2the hydrophone converts the received acoustic signals into electric signals, and the electric signals enter the DSP through the A/D converter after being amplified and subjected to band-pass filtering; the frequency range of the band-pass filter is set to (f)1-W1/2)～(f1+W1/2+W2) Wherein f is1For sub-band B1The center frequency of (d);
7)U2the subtask 2 of the DSP performs synchronous signal detection on the signals obtained in the step 6); after synchronization, estimating Doppler frequency shift by detecting the change of the time difference of the two frequency modulation signals, and then performing Doppler frequency shift compensation on the received signals by adopting a linear interpolation method;
8)U2the subtask 2 of the DSP carries out FFT inverse transformation on the signal obtained in the step 7), and according to the FFT transformed value Y of the 1 st OFDM symbol1(k) And pilot symbols s (k) estimating the underwater acoustic channel transfer function as:
wherein, K1For sub-band B1The number of subchannels; n isf＝Nf1/Fs；FsIs the sampling frequency, and N is the FFT transform length; estimating the received symbol of each subchannel of the 2 nd OFDM symbol according to the formula (1) as follows:
the noise power estimate for the 2 nd OFDM symbol is:
wherein, Y2(k) FFT transform for the 2 nd OFDM symbol; n is2＝Nf2/Fs，f2For sub-band B2The center frequency of (d); k2For sub-band B2The number of subchannels; to pair
Performing symbol demapping, and inputting the generated soft information and the noise power w into a channel decoder; the channel decoder outputs code words to generate s '(k) after symbol mapping, the s' (k) is fed back to the underwater sound channel estimator, and the new underwater sound channel transfer function is estimated as follows:
equalizing and demodulating symbols of each subchannel of the 3 rd OFDM symbol according to a formula (4); according to the same method, until all the voice data transmitted by n-1 OFDM symbols are demodulated; the output information of the channel decoder is sent to the vocoder through an RS232 serial port after hard decision;
9) the vocoder decodes the voice coded data obtained in the step 8), and the generated voice signal is played through a loudspeaker (or an earphone);
10)U2voice signal communication process picked up by microphone and U1Identical except that U2The sub-band used for transmitting speech data is B3Thus, U1The corresponding band-pass filter has a frequency range of (f)3-W1/2-W2)～(f3+W1/2) wherein nf＝Nf3/Fs，f3For sub-band B3The center frequency of (d);
11) speech signal slave U1Is transmitted to U2And a slave U2Is transmitted to U1The communication process of the system is completely independent, and the full-duplex communication function is realized.
The invention relates to a real-time full-duplex underwater acoustic digital voice communication system and a method thereof based on a vocoder and a digital signal processor, which combine a low-rate voice compression coding and decoding technology and an orthogonal frequency division multiplexing multi-carrier modulation communication technology. Compared with the prior art, the invention has the following outstanding advantages:
1. bandwidth division of underwater acoustic transducer into B1，B2And B3Equal 3 different sub-bands, sub-band B1And B3For transmitting speech data, sub-band B2For estimating the noise power. Speech signal slave U1Is transmitted to U2And a slave U1Is transmitted to U2The process of the method is completely independent, and the real-time full-duplex communication function is realized.
2. Advanced low-rate voice compression coding, OFDM multi-carrier communication, self-adaptive channel estimation and tracking technology are adopted, the system time delay is small, the system adapts to a rapid time-varying underwater acoustic channel environment, and the voice synthesized by a receiving end has high intelligibility, naturalness and definition.
3. The system is portable, stable, low in power consumption and low in price. The voice compression coding and decoding adopts a small-sized vocoder chip, and core algorithms of signal synchronization, Doppler frequency shift estimation and compensation, adaptive channel estimation and equalization, modulation/demodulation, channel coding/decoding and the like are all completed in one DSP chip.
4. The system is scalable, easy to debug and high in maintainability.
Drawings
Fig. 1 is a block diagram of a full-duplex underwater acoustic digital voice communication system.
Fig. 2 shows WT600F vocoder module.
Fig. 3 is a schematic diagram of a preamplifier.
Fig. 4 is a schematic diagram of a bandpass filter.
Fig. 5 is a schematic diagram of a power amplifier and matcher.
FIG. 6 is a schematic diagram of a power management module.
Fig. 7 is a flow chart of the DSP program.
Fig. 8 is a transmit signal frame structure.
Fig. 9 is an underwater acoustic channel estimation and equalization architecture.
Detailed Description
The following examples will further illustrate the present invention with reference to the accompanying drawings.
The full-duplex underwater acoustic digital voice communication system is shown in fig. 1. The system is divided into two parties of communication A and B, and recorded as U1And U2。U1And U2The system has the same hardware structure and mainly comprises a microphone, earphones, a WT600F vocoder, a TMS 320C 6748DSP, a power amplification and matching device, a preamplifier, a band-pass filter, a circuit management module, an underwater acoustic transducer, a hydrophone and other components. The underwater acoustic transducer has the bandwidth of 20-30 kHz and is divided into 3 sub-bands of 20-24 kHz, 24-26 kHz and 26-30kHz, and a communication party U1Using a 20-24 kHz sub-band to transmit signals, and using a 26-30kHz sub-band to receive signals; communication side U2And transmitting signals by using a sub-band of 26-30kHz, receiving signals by using a sub-band of 20-24 kHz, and estimating noise power by using a sub-band of 24-26 kHz. The communication process is briefly described as follows: communication side U1Voice signals picked up by the microphone are coded by a WT600F vocoder and then sent to the C6748DSP through an RS232 serial port; the DSP extracts effective voice coding data, and generates a baseband transmission signal after LDPC coding, pilot symbol insertion, IFFT conversion and framing are carried out on the effective voice coding data; frequency shifting of a baseband transmission signal to a frequency range of 20-24 kHz of a transmitting sub-band, and generation of a transmitting signal through D/A conversion; the transmit signal excites the underwater acoustic transducer to radiate acoustic waves propagating in the underwater acoustic channel. Communication side U2The hydrophone converts the received acoustic signal into an electric signal; the electric signals are filtered and amplified by an analog band-pass filter with a passband of 20-26 kHz and then enter the DSP; the DSP performs synchronous detection, Doppler frequency shift estimation and compensation, FFT conversion, channel estimation and equalization and LDPC decoding on the received signal to generate received data; the DSP frames the received data in a voice encoded data format and sends the framed data to WT600F in a timed manner through an RS232 serial port; the voice signal synthesized by WT600F is output through headphones or earphones. Communication side U2The communication process of the voice signals picked up by the microphone is substantially the same as the above process, except that U2The sub-band of the transmitted signal is 26-30kHz, and correspondingly, the passband of the first-square band-pass filter is set to be 24-30 kHz. The serial communication parameters between the vocoder and the DSP are as follows: baud rate 115200bps, data bit 8, stop bit 1, no check.
The WT600F vocoder is a low-rate vocoder chip, as shown in FIG. 2. The WT600F vocoder has built-in voice codec software, and can realize the compression and synthesis of voice at the same time without external memory, and synthesize and output voice with higher quality at 600bps rate. The WT600F vocoder provides a UART interface through which a user may read and write vocoded data. The format of data read and written by the WT600F vocoder is shown in Table 1. In table 1, the 5 th to 9 th bytes and the high 5 of the 10 th byte are speech coded data, and the rest bytes are frame header, command symbol, length, CRC check, etc., so that the effective speech coded data is 45 bits.
Table 1 WT600F vocoder frame structure
| Header_1 | Head_2 | CMD | LEN | DATA | CRC |
| B1 | B2 | B3 | B4 | B5-B15 | B16 |
The preamplifier adoptsA two-stage in-phase proportional amplifying circuit formed by an operational amplifier OPA227 of the masterpiece, as shown in fig. 3. The gain bandwidth product of the OPA227 is 8MHz, and the gain bandwidth product has extremely low noise, extremely low drift and extremely high precision, the open loop gain is more than 140dB, the output capacity is 50mA, the current protection is carried out, the burning-out is not easy, and the direct current and alternating current characteristics are extremely good. The in-phase proportional amplifying circuit can change the amplification factor by changing the resistance values of two resistors connected with the cathode of the input of the amplifier. The first stage magnification is 11 times, the second stage magnification is 26 times, and the total magnification can reach nearly 300 times. The second-stage feedback resistor adopts a potentiometer, so that the gain of the system can be adjusted to achieve the optimal output voltage, and the post-stage filter can conveniently process the voltage. The band-pass filter is Chebyshev filter and has passband bandwidth range U 124 to 30kHz and 20 to 26kHz for the second square. The other parameters are as follows: the passband gain is 0dB, the maximum attenuation is-2 dB, and the passband ripple is 0.01 dB; the stop band bandwidth is 17kHz and the attenuation is-20 dB. Finally, a low-noise precision operational amplifier ADA4004-4 is dazzled to build the filter, and a schematic diagram of the filter is shown in FIG. 4.
The power amplifier and matcher are shown in fig. 5. The power amplifier adopts a TPA3118 audio chip, can provide maximum 50W power for a single-channel load of 8 ohms under the voltage of 24V, and the efficiency is higher than 90%. The switching frequency of the TPA3118 is up to 1.2MHz after setting, so that AM interference can be effectively avoided. Meanwhile, the TPA3118 chip integrates a self-protection circuit, and the circuit comprises overvoltage, undervoltage, over-temperature, direct current detection and short circuit, so that burning can be effectively avoided. An inductor is connected in parallel between the output of the power amplifier and the input of the underwater acoustic transducer, so that matching between the transducer and the power amplifier impedance is realized, and the acoustic power radiated by the underwater acoustic transducer to the water medium is increased.
The underwater sound digital voice communication system is powered by a 24V lithium battery, stable 24V and +/-5V direct-current voltages need to be provided, the former supplies power for a power amplifier and a matcher, and the latter supplies power for a WT600F vocoder, a C6748DSP, a preamplifier, a band-pass filter and the like. The 24V to +/-5V DC-DC power conversion module adopts a TPS5430 wide input step-down chip of TI company, the highest output current can reach 3A, the conversion efficiency is as high as 95%, and a specific circuit is shown in fig. 6 (a). An XL4015 voltage stabilization chip is adopted for 24V voltage stabilization, and a specific circuit is shown in FIG. 6 (b). XL4015 can realize the adjustment of voltage through changing the size of two resistances, and the highest heavy current of exportable 5A is particularly suitable for high-power amplifier circuit to use.
Except that the speech compression codec is implemented in the WT600F vocoder, the remaining procedures and algorithms of the full-duplex underwater acoustic digital speech communication system, including signal synchronization, doppler shift estimation and compensation, modulation/demodulation, channel estimation and equalization, LDPC codec, etc., are implemented in one DSP. The program flow is shown in fig. 7, and includes two subtasks in addition to the main program, where the subtask 1 implements the sending process of the voice data, and the subtask 2 implements the receiving process of the voice data. Fig. 8 is a frame structure of a transmission signal, a section of identical frequency modulation signals are inserted before and after each frame signal, the doppler frequency shift of a channel can be estimated by detecting the time interval change of the two frequency modulation signals, and the doppler frequency shift compensation is performed accordingly. In fig. 8, the 1 st OFDM symbol transmits pilot symbols, and the remaining 3 OFDM symbols transmit voice data. Fig. 9 is an underwater acoustic channel estimation and equalization architecture. Let the output symbol after FFT, i.e. the channel input symbol in FIG. 9, be Yi(k)(i＝1,2,3,4,k＝nf-K1/2,…,nf,…,nf+K1/2-1), for U1，nf＝Nf3/FsTo U, to U2，nf＝Nf1/FsWherein f is1＝22kHz，f328 kHz. Channel transfer function H corresponding to each OFDM symboli(k) Estimated as:
wherein
Wherein s (k) is a pilot symbol, and s' (k) is a symbol which is output by the LDPC decoder, symbol-mapped, and fed back to the channel estimator. The underwater acoustic channel transfer function can be considered to remain unchanged between two consecutive OFDM symbols, and the received symbol is estimated as:
Claims (2)
1. A full-duplex underwater sound digital voice communication system is characterized by consisting of a communication party A and a communication party B which are respectively marked as U1And U2(ii) a Communication side U1And communication party U2The positions of the voice signal receiving and transmitting units are completely the same, and the voice signal receiving and transmitting units can independently and autonomously send and receive voice signals, so that the full-duplex voice communication function is realized; the underwater acoustic transducer adopts a round tube type broadband transducer, and the bandwidth of the transducer is divided into B1、B2And B33 different sub-bands, of which sub-band B1And B3Having the same bandwidth W1，B2Has a bandwidth of W2(ii) a To communication party U1，B1For the uplink sub-band, for transmitting speech data, B2Is a downlink sub-band for receiving voice data; to U2Then, in contrast, B2For the uplink sub-band, for transmitting speech data, B1Is a downlink sub-band for receiving voice data; sub-band B3For estimating the noise power;
U1and U2The hardware part of the system is composed of voice input/output equipment, vocoder and its peripheral circuit, DSP and its peripheral circuit, power amplifier and its matching circuit, preamplifier, band-pass filter and electric circuitThe system comprises a source management module, an underwater acoustic transducer, a hydrophone and other components; the voice input and output device is used for picking up and playing voice, and the vocoder realizes parameter compression coding and decoding of the voice; the DSP comprises two subtasks besides a main program, wherein the subtask 1 realizes transmitting underwater sound signal processing including channel coding, OFDM modulation and the like, and the subtask 2 realizes receiving underwater sound signal processing including functions of synchronization, Doppler frequency shift estimation and compensation, self-adaptive channel estimation and equalization, channel decoding and the like; each frame signal comprises 1 synchronization head and n OFDM symbols, wherein the 1 st OFDM symbol is transmitted by pilot symbols, and the 2 nd to nth OFDM symbols are transmitted by voice data symbols.
2. A full duplex underwater sound digital voice communication method is characterized by comprising the following steps:
1)U1inputting the voice signal picked up by the microphone into a low-speed vocoder, performing parameter compression coding on the voice signal by the vocoder, and extracting voice coding data according to frames;
2)U1the vocoder transmits the voice coded data obtained in the step 1) to the DSP through an RS232 serial port;
3)U1a subtask 1 of the DSP performs framing, channel coding, symbol mapping, pilot symbol insertion, IFFT conversion and other baseband signals on the voice data obtained in the step 2), and inserts a section of frequency modulation signals with the same length and start-stop frequency before and after each frame signal; each frame signal comprises n OFDM symbols, the 1 st OFDM symbol is all pilot symbols, and the rest n-1 OFDM symbols are all voice data symbols;
4)U1the subtask 1 of the DSP shifts the baseband signal obtained in the step 3) to a sub-band B1The frequency range of (d);
5)U1the DSP subtask 1 outputs the signal obtained in the step 4) to a power amplifier through a D/A converter, and the electric signal after power amplification excites an underwater acoustic transducer to emit sound waves to be transmitted in an underwater acoustic channel;
6)U2the hydrophone converts the received acoustic signal into electric signal, which is amplified and band-pass filtered and converted into A/D signalThe converter enters a DSP; the frequency range of the band-pass filter is set to (f)1-W1/2)～(f1+W1/2+W2) Wherein f is1For sub-band B1The center frequency of (d);
7)U2the subtask 2 of the DSP performs synchronous signal detection on the signals obtained in the step 6); after synchronization, estimating Doppler frequency shift by detecting the change of the time difference of the two frequency modulation signals, and then performing Doppler frequency shift compensation on the received signals by adopting a linear interpolation method;
8)U2the subtask 2 of the DSP carries out FFT inverse transformation on the signal obtained in the step 7), and according to the FFT transformed value Y of the 1 st OFDM symbol1(k) And pilot symbols s (k) estimating the underwater acoustic channel transfer function as:
wherein, K1For sub-band B1The number of subchannels; n isf＝Nf1/Fs；FsIs the sampling frequency, and N is the FFT transform length; estimating the received symbol of each subchannel of the 2 nd OFDM symbol according to the formula (1) as follows:
the noise power estimate for the 2 nd OFDM symbol is:
wherein, Y2(k) FFT transform for the 2 nd OFDM symbol; n is2＝Nf2/Fs，f2For sub-band B2The center frequency of (d); k2For sub-band B2The number of subchannels; to pair
Performing symbol demapping, and inputting the generated soft information and the noise power w into a channel decoder; the channel decoder outputs code words to generate s '(k) after symbol mapping, the s' (k) is fed back to the underwater sound channel estimator, and the new underwater sound channel transfer function is estimated as follows:
equalizing and demodulating symbols of each subchannel of the 3 rd OFDM symbol according to a formula (4); according to the same method, until all the voice data transmitted by n-1 OFDM symbols are demodulated; the output information of the channel decoder is sent to the vocoder through an RS232 serial port after hard decision;
9) the vocoder decodes the voice coded data obtained in the step 8), and the generated voice signal is played through a loudspeaker (or an earphone);
10)U2voice signal communication process picked up by microphone and U1Identical except that U2The sub-band used for transmitting speech data is B3Thus, U1The corresponding band-pass filter has a frequency range of (f)3-W1/2-W2)～(f3+W1/2) wherein nf＝Nf3/Fs，f3For sub-band B3The center frequency of (d);
11) speech signal slave U1Is transmitted to U2And a slave U2Is transmitted to U1The communication process of the system is completely independent, and the full-duplex communication function is realized.
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