CN114157371A - Low-power-consumption underwater digital voice communication method and communication system - Google Patents
Low-power-consumption underwater digital voice communication method and communication system Download PDFInfo
- Publication number
- CN114157371A CN114157371A CN202111346209.XA CN202111346209A CN114157371A CN 114157371 A CN114157371 A CN 114157371A CN 202111346209 A CN202111346209 A CN 202111346209A CN 114157371 A CN114157371 A CN 114157371A
- Authority
- CN
- China
- Prior art keywords
- signal
- module
- voice
- transmitting
- digital
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Pending
Links
- 238000004891 communication Methods 0.000 title claims abstract description 64
- 238000000034 method Methods 0.000 title claims abstract description 30
- 238000012545 processing Methods 0.000 claims abstract description 37
- 230000006835 compression Effects 0.000 claims abstract description 33
- 238000007906 compression Methods 0.000 claims abstract description 33
- 238000006243 chemical reaction Methods 0.000 claims abstract description 24
- 238000001228 spectrum Methods 0.000 claims abstract description 13
- 238000005070 sampling Methods 0.000 claims description 18
- 230000005284 excitation Effects 0.000 claims description 14
- 230000000737 periodic effect Effects 0.000 claims description 11
- 230000001360 synchronised effect Effects 0.000 claims description 11
- 230000003321 amplification Effects 0.000 claims description 8
- 230000015572 biosynthetic process Effects 0.000 claims description 8
- 238000003199 nucleic acid amplification method Methods 0.000 claims description 8
- 238000003786 synthesis reaction Methods 0.000 claims description 7
- 239000013078 crystal Substances 0.000 claims description 6
- 238000001914 filtration Methods 0.000 claims description 6
- 238000004321 preservation Methods 0.000 claims description 6
- 238000013139 quantization Methods 0.000 claims description 6
- 230000003595 spectral effect Effects 0.000 claims description 6
- 238000012360 testing method Methods 0.000 claims description 6
- 230000007480 spreading Effects 0.000 claims description 4
- 238000005311 autocorrelation function Methods 0.000 claims description 3
- 238000012937 correction Methods 0.000 claims description 3
- 238000000605 extraction Methods 0.000 claims description 3
- 238000013507 mapping Methods 0.000 claims description 3
- 238000007781 pre-processing Methods 0.000 claims description 3
- 230000008569 process Effects 0.000 claims description 3
- 239000000126 substance Substances 0.000 claims description 3
- 230000000694 effects Effects 0.000 abstract description 4
- 238000010586 diagram Methods 0.000 description 6
- 230000005540 biological transmission Effects 0.000 description 5
- 230000007547 defect Effects 0.000 description 5
- 238000005516 engineering process Methods 0.000 description 3
- 238000011160 research Methods 0.000 description 3
- 230000002087 whitening effect Effects 0.000 description 2
- 241000282414 Homo sapiens Species 0.000 description 1
- 230000001133 acceleration Effects 0.000 description 1
- 230000001427 coherent effect Effects 0.000 description 1
- 238000013461 design Methods 0.000 description 1
- 238000011161 development Methods 0.000 description 1
- 230000009189 diving Effects 0.000 description 1
- 230000006870 function Effects 0.000 description 1
- 238000001208 nuclear magnetic resonance pulse sequence Methods 0.000 description 1
- 230000005236 sound signal Effects 0.000 description 1
- 230000004083 survival effect Effects 0.000 description 1
- 230000002194 synthesizing effect Effects 0.000 description 1
- 238000012549 training Methods 0.000 description 1
- 230000009466 transformation Effects 0.000 description 1
- XLYOFNOQVPJJNP-UHFFFAOYSA-N water Substances O XLYOFNOQVPJJNP-UHFFFAOYSA-N 0.000 description 1
Images
Classifications
-
- 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
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; 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
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; 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
- G10L19/173—Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding
Landscapes
- Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Computational Linguistics (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Computer Networks & Wireless Communication (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
The invention provides a low-power-consumption underwater digital voice communication method and a communication system.A power supply management module provides different required power supply voltages, a voice compression coding and decoding module is connected with a signal processing and storing module through a serial port, the signal processing and storing module is connected with a transmitting and receiving module through an A/D conversion module and a D/A conversion module, and the power supply management module introduces direct-current voltage from the outside to realize the management of input voltage. The invention has low power consumption and high efficiency, and realizes the maximum electroacoustic conversion efficiency; meanwhile, the system has strong concealment, the power in the time domain is reduced by adopting low transmitting power, the frequency spectrum is widened in a direct sequence spread spectrum mode, the total energy is unchanged, and the power in the frequency domain is reduced; in addition, the system has high reliability, and the direct sequence spread spectrum mode has stronger anti-multipath interference capability and can achieve good real-time voice communication effect at lower communication speed.
Description
Technical Field
The invention relates to the technical field of underwater acoustic communication, in particular to an underwater voice communication method and an underwater voice communication system.
Background
The underwater voice communication has wide application in many aspects such as marine scientific research, marine resource exploration, frogman diving, submarine and ship military training and the like, but the underwater voice communication technology is not mature and still is in a starting stage. With the acceleration of the development of ocean pace of human beings, the research of underwater voice communication technology is increasingly paid attention by people.
The traditional underwater voice communication adopts a wired mode, the distortion of the voice output by a receiving end is small, and the receiving end has high intelligibility, naturalness and definition, but the wired mode has the defects of small moving range, easy formation of stranded cables and the like, so the application range is greatly limited. The application range can be expanded by adopting a wireless communication mode, but the underwater wireless voice communication by utilizing sound waves is one of the most difficult and challenging research subjects in the ocean technology, which is mainly determined by the limited communication bandwidth of an underwater sound channel and the characteristics of time variation, space variation and the like. The underwater wireless voice communication mainly comprises two modes of analog single-side band modulation and digital modulation. The analog modulation communication mode is greatly influenced by the marine underwater acoustic environment, the influence of time-varying multipath and time-varying Doppler in a shallow sea channel is difficult to overcome, the voice communication quality is difficult to guarantee, and the system has the defects of large size, low power utilization rate, easy crosstalk among different users and the like.
The digital voice communication can overcome the defects of analog voice communication and restore higher voice quality. However, the digital voice communication is performed by using the sound wave underwater, because the available communication bandwidth of the underwater sound channel is limited and the communication rate is low, the voice information must be greatly compressed to reduce the bit rate required by transmission. In addition, the quality of the acoustic signal is also easily influenced by space-time complex ocean channels, and is easily intercepted by enemy sonar reconnaissance equipment to reduce the survival degree of underwater targets. Moreover, the long propagation delay due to the multipath effect of shallow sea underwater sound broadens the signal, and the inter-symbol interference caused thereby reduces the reliability of communication. Therefore, how to realize low-power consumption, high-efficiency, reliable and concealed digital voice transmission under water becomes a key technical problem to be solved urgently at present.
Disclosure of Invention
In order to overcome the defects of the prior art, the invention provides a low-power-consumption underwater digital voice communication method and a communication system. Aiming at the problems, the invention aims to overcome the defects of the existing underwater voice communication system and method and aims to provide an underwater sound digital voice communication system and method with low power consumption, high efficiency, high intelligibility, good naturalness, strong reliability, strong concealment and low speed.
The technical scheme adopted by the invention for solving the technical problem specifically comprises the following steps:
the method comprises the following steps: MELP compression encoding of input speech signals
Inputting an analog voice signal x (t) into a microphone at a transmitting end by a vocoder, converting the analog voice signal x (t) through A/D, carrying out low-rate compression coding on the converted digital voice through an improved low-rate voice compression coding algorithm MELP to obtain a compressed digital voice signal a (t), wherein the determination of a pitch period and a non-periodic voiced section of the MELP algorithm is the key of the algorithm;
step two: modulating compressed digital speech signals
Carrying out symbol mapping on the obtained compressed digital signal a (t), and preparing a modulation signal s (t) by adopting direct sequence spread spectrum and BPSK modulation;
the symbol rate of the compressed digital signal a (t) is RaC (t) is a pseudo-random sequence with a rate RcThe spreading process is a modulo-quadratic multiplication of a (t) and a pseudo-random sequence c (t), and the spread sequence d (t) is expressed as:
d(t)＝a(t)c(t)
moving d (t) to the carrier frequency for BPSK modulation, and the modulated signal s (t) is expressed as:
s(t)＝d(t)cosωct＝a(t)c(t)cosωc(t)；
step three: sampling the modulation signal to generate PWM wave and then transmitting;
sampling a modulation signal s (t) and a triangular wave t (n) to generate a PWM wave, generating a high-frequency triangular wave by utilizing DSP programming, simultaneously sampling the modulation signal and the triangular wave by using the same sampling rate, wherein the amplitude of the modulation signal is S (n), the amplitude of the triangular wave is T (n), comparing the amplitude of the modulation signal with the amplitude of the triangular wave, and outputting a high level if S (n) is more than or equal to T (n); if S (n) < T (n), outputting a low level;
the PWM wave generated by sampling is converted into sound wave through a D/A converter, a power amplifier, a transformer and a matching network, and finally the sound wave is converted into the sound wave through an underwater acoustic transducer to be transmitted in an underwater acoustic channel;
step four: preprocessing a received signal;
after the signal transmitted in the third step is transmitted through an underwater acoustic channel, the transducer receives and converts the transmitted weak acoustic signal into an electric signal, the electric signal is amplified in a front-mounted triode, AGC (automatic gain control) is carried out on an automatic gain control chip, a band-pass filter is carried out on the electric signal to obtain a bipolar analog signal, the analog signal is subjected to FFT (fast Fourier transform) conversion after A/D (analog to digital) conversion, and at the moment, the preprocessed received signal r (t) starts synchronous identification;
step five: demodulating a pre-processed received signal
Despreading and BPSK demodulation are carried out on the preprocessed received signals r (t) to obtain demodulation signals r' (t);
the signal received by the receiver is selectively amplified and mixed to obtain a useful signal sI(t) channel noise nI(t) and the interference signal, i.e. the mixed signal is:
rI(t)＝sI(t)+nI(t)+JI(t)
the receiving end generates a pseudo-random sequence c '(t) which is the same as the transmitting end, and the received signal is multiplied by the local pseudo-random sequence c' (t) for despreading:
rI'(t)＝rI(t)c'(t)＝sI'(t)+nI'(t)+JI'(t)
wherein s isI'(t)＝sI(t)c'(t)＝a(t)c(t)c'(t)cosωc(t)＝a(t)cosωc(t)；
Noise separationQuantity nI' (t) is Gaussian band-limited white noise, the spectral density is basically unchanged but the relative bandwidth is changed after demodulation, the noise power is reduced, and the interference component JI' (t) is uncorrelated with the pseudo-random code, the spectral density is reduced, and the interference power entering the demodulation is only the same part of the signal frequency band;
step six: performing MELP decoding on the demodulated signal and playing
Carrying out MELP decoding on the signal r' (t) processed in the step five by using an improved low-rate speech compression coding algorithm to synthesize a digital speech signal, reducing the digital speech signal into an analog speech signal y (t) by using a vocoder D/A converter, and playing the analog speech signal y (t) by using an earphone or a loudspeaker;
the generation of the excitation signal is a key part of MELP decoding, and in one pitch period, the excitation signal is obtained through IDFT operation:
wherein T is a pitch period, which is a sum of an interpolated pitch value and a product of a pitch shake intensity and a pitch, and M (k) (k is 1, 2.. 10) is an interpolated fourier series amplitude value, and M (T-k) is M (k);
and (3) voice synthesis, namely, enabling the excitation signal to pass through a synthesis filter, and then carrying out gain correction, pulse discrete filtering and post-filtering treatment to finally obtain an analog voice signal y (t).
In the first step, the pitch period first defines a normalized autocorrelation function r (τ):
wherein the content of the first and second substances,
xk+m、xk+nrespectively representing the k + m and k + n sampling points after the analog voice signal x (t) generates a digital signal through A/D conversion, wherein m and n belong to (0, tau), the tau value range is 40-160, and the autocorrelation r (tau) value is largeWhen the pitch period is equal to 0.6, the final pitch period can be determined;
and (3) the non-periodic voiced sound segment is determined by sub-band sound intensity extraction and quantization, and the peak value of the residual signal is calculated:
wherein r isnFor residual signals, if the peak a is greater than 1.34, the first sub-band is voiced, and if the peak is greater than 1.6, the first three sub-bands are voiced;
and quantizing according to the quantization codebook according to the generated pitch period and the determined non-periodic voiced segment to generate a compressed digital signal a (t).
In the fourth step, the step of synchronous identification is as follows: performing cross-correlation operation on the data and the m sequence by adopting an overlap preservation method, adopting an overlap preservation method of continuous 17 windows when synchronizing the head, adopting an overlap preservation method of continuous 3 windows when synchronizing the data, calculating the maximum peak value in each window, and solving the data initial position corresponding to the maximum peak value to finish synchronous identification;
the invention also provides a low-power-consumption underwater digital voice communication system, which comprises a power supply management module, a voice compression coding and decoding module, a signal processing and storing module and a transmitting and receiving module; the power management module provides different required power supply voltages for the voice compression coding and decoding module, the signal processing and storage module and the transmitting and receiving module, the voice compression coding and decoding module is connected with the signal processing and storage module through a serial port, the signal processing and storage module is connected with the transmitting and receiving module through an A/D conversion module and a D/A conversion module, and when the communication system is in a transmitting state, an input voice signal is transmitted after sequentially passing through the voice compression coding module, the signal processing and storage module and the transmitting module; when the communication system is in a receiving state, the received signals are played after passing through a receiving module, a signal processing and storing module and a voice compression decoding module respectively; the power management module introduces 18-28V direct current voltage from the outside to realize the management of input voltage.
The voice compression coding and decoding module comprises a microphone, a microphone output signal A/D acquisition circuit, a vocoder signal D/A converter, an earphone or a loudspeaker, when the system is in a transmitting state, after voice is input into the microphone, the output end of the microphone is connected with the input end for the A/D acquisition of the microphone output signal, the output end for the A/D acquisition of the microphone output signal is connected with the input end of the vocoder, and the output end of the vocoder is connected with the DSP input end of the signal processing and storage module; when the system is in a receiving state, the DSP output end of the signal processing and storing module is connected with the input end of the vocoder signal D/A converter, and the output end of the vocoder signal D/A converter is connected with an earphone or a loudspeaker.
The signal processing and storing module comprises a Digital Signal Processor (DSP), a serial port communication chip, a digital signal D/A converter, an analog signal A/D converter, a flash and a crystal oscillator circuit, wherein the crystal oscillator circuit provides a clock for the DSP, and the flash is a program for testing a DSP storing system; when the system is in a transmitting state, the output end of a voice coder and decoder module vocoder is input into the DSP through a serial port communication chip MAX3223, the DSP outputs the processed voice to the input end of a digital signal D/A converter, and the output end of the digital signal D/A converter is connected with the input end of a power amplifier of a transmitting and receiving module; when the system is in a receiving state, the output end of the transmitting and receiving module band-pass filter is connected with the input end of the analog signal A/D converter, the output end of the analog signal A/D converter is connected with the input end of the DSP, and the output end of the DSP is connected with the input end of the voice coder-decoder module vocoder signal D/A converter.
The transmitting and receiving module comprises a power amplifier, a transformer, a matching network, an underwater acoustic transducer, a pre-stage amplifier, an automatic gain control AGC and a band-pass filter, when the system is in a transmitting state, the output end of a digital signal D/A converter of the signal processing and storing module is connected with the input end of the power amplifier, the output end of the power amplifier is connected with the input end of the transformer and the matching network, the output end of the transformer and the matching network is connected with the underwater acoustic transducer, and the underwater acoustic transducer transmits signals to an underwater acoustic channel; when the system is in a receiving state, the underwater acoustic transducer inputs a received signal into a triode for pre-amplification, the output end of the pre-amplification is connected with the input end of the automatic gain control AGC, the output end of the automatic gain control AGC is connected with the input end of the band-pass filter, and the output end of the band-pass filter is connected with the input end of the analog signal A/D converter of the signal processing and storage module.
The invention has the advantages that the system has low power consumption and high efficiency, the AP280 vocoder designed with low power consumption is adopted, the power control module designed at the transmitting end can select the system power according to the requirement, the processing of modulating, demodulating, spreading, despreading, channel coding, decoding and the like of the underwater sound signal are realized in one DSP chip, the DC-DC module adopted by the power supply design achieves 85 percent of conversion efficiency, the D-type power amplifier with high conversion efficiency is adopted by the power amplifier at the transmitting end, and the impedance matching of the transmitting signal and the transducer is realized by the transformer and the matching network, thereby realizing the maximum electroacoustic conversion efficiency; meanwhile, the system has strong concealment, the power in the time domain is reduced by adopting low transmitting power, the frequency spectrum is widened in a direct sequence spread spectrum mode, the total energy is unchanged, and the power in the frequency domain is reduced; in addition, the system has high reliability, the receiving end adopts automatic gain control AGC to effectively adapt to reliable synchronization of communication at different distances, the direct sequence spread spectrum mode has stronger anti-multipath interference capability, and can achieve good real-time voice communication effect at lower communication speed.
Drawings
FIG. 1 is a schematic block diagram of the principle of the low-power-consumption underwater digital voice communication method of the present invention.
Fig. 2 is a schematic diagram of a frame structure of a transmission signal according to the present invention.
Fig. 3 is a block diagram of the low power consumption underwater digital voice communication system of the present invention.
Fig. 4 is a waveform diagram of a speech signal of a pond test according to the present invention, in which (a) is original speech and (b) is speech synthesized by a communication receiver after passing through an AP280 vocoder.
Detailed Description
The invention is further illustrated with reference to the following figures and examples.
The low-power consumption underwater sound voice communication method comprises the following specific steps:
the method comprises the following steps: MELP compression encoding of input speech signals
The invention adopts a Mixed Excitation Linear Prediction (MELP) algorithm, the MELP algorithm is based on the traditional linear prediction coding parameter model and combines the ideas of multi-band excitation and mixed excitation, the natural voice characteristics are better simulated, and higher voice quality can be restored under the condition of low speed. The MELP algorithm has a sampling rate of 8kHz, each frame time interval is 22.5ms, one frame comprises 180 voice samples, the algorithm divides the voice into three states of unvoiced sound, voiced sound and jittered voiced sound, the unvoiced sound adopts white noise as an excitation signal, the voiced sound adopts periodic pulse sequence whitening noise with the period equal to the pitch period as an excitation signal, and the jittered voiced sound adopts non-periodic pulse whitening noise as a synthesized excitation signal. Therefore, the determination of pitch periods and non-periodic voiced segments is the key to the algorithm.
At the transmitting end, a vocoder is used to perform low-rate compression coding on the analog voice signal x (t) input by the microphone after A/D conversion by using an improved low-rate voice compression coding algorithm MELP to obtain a compressed digital voice signal a (t). The determination of pitch periods and non-periodic voiced segments of the MELP algorithm is the key to the algorithm.
The pitch period first defines a normalized autocorrelation function r (τ):
wherein the content of the first and second substances,
xk+m、xk+nrespectively representing the k + m sampling points and the k + n sampling points after the analog voice signal x (t) generates a digital signal through A/D conversion, wherein m and n belong to (0, tau), the tau value range is 40-160, and the final pitch period can be determined when the autocorrelation value is more than or equal to 0.6.
And (3) the non-periodic voiced sound segment is determined by sub-band sound intensity extraction and quantization, and the peak value of the residual signal is calculated:
whereinrnFor residual signals, the first sub-band is voiced if the peak a is greater than 1.34, and the first three sub-bands are voiced if the peak is greater than 1.6.
And quantizing according to the quantization codebook according to the generated pitch period and the determined non-periodic voiced segment to generate a compressed digital signal a (t).
One implementation of the MELP speech compression coding algorithm is to employ an AP280 vocoder. The AP280 is a low-bit-rate voice coding and decoding chip, is internally provided with CODEC and voice coding and decoding software, does not need an external memory, and can simultaneously realize the compression coding and the synthesis of voice; the standard UART interface is adopted to be connected with the MCU, and the reading and writing of the voice coded data can be realized through the UART interface; and the LQFP48 package is adopted to support low power consumption and sleep mode.
Step two: modulating compressed digital speech signals
And carrying out symbol mapping on the obtained compressed digital voice signal a (t), and preparing a modulation signal s (t) by adopting direct sequence spread spectrum and BPSK modulation.
The symbol rate of the compressed digital signal a (t) is RaC (t) is a pseudo-random sequence with a rate RcThe spreading process is a modulo-quadratic multiplication of a (t) and the pseudo-random sequence c (t), and the spread sequence d (t) is expressed as:
d(t)＝a(t)c(t)
moving d (t) to the carrier frequency for BPSK modulation, and the modulated signal s (t) is expressed as:
s(t)＝d(t)cosωct＝a(t)c(t)cosωc(t)
the format of the final transmit one frame data structure is shown in fig. 2. Firstly, transmitting a synchronous head, then, transmitting blank interval, then, transmitting data, wherein a certain protection interval is needed between a frame synchronous signal and the data, and the length of the protection interval is larger than the maximum time delay expansion of a channel so as to prevent the multipath effect of the frame synchronous signal from influencing the data. The direct sequence spread spectrum processing is carried out on the digital voice data by using the PN code, the direct sequence spread spectrum adopts a pseudo-random code m sequence mode to spread the frequency spectrum of a transmitting signal at a transmitting end, wherein the pseudo-random code m sequence has good autocorrelation and is independent with a transmitting code element, the original information code can be restored by carrying out coherent de-spread only by using the same code element at a receiving end, and when the multipath time delay in underwater acoustic communication exceeds the length of one spread spectrum code, the correlation between the information code and the original code element is sharply reduced, so that the strong anti-multipath interference capability is realized.
Step three: generating PWM wave emission by sampling modulation signal
Sampling a modulation signal s (t) and a triangular wave t (n) to generate a PWM wave, generating a high-frequency triangular wave by utilizing DSP programming, simultaneously sampling the modulation signal and the triangular wave by using the same sampling rate, wherein the amplitude of the modulation signal is S (n), the amplitude of the triangular wave is T (n), comparing the amplitude of the modulation signal with the amplitude of the triangular wave, and outputting a high level if S (n) is more than or equal to T (n); if S (n) < T (n), a low level is outputted. At the moment, bipolar PWM waves are formed, analog signals are input to a D-class power amplifier for amplification through D/A conversion, the theoretical conversion rate of the D-class power amplifier can reach 100%, the actual efficiency is higher than 80%, and the conversion rate is 2-3 times that of a traditional linear power amplifier. The power amplification circuit designed by the invention simultaneously supports half-bridge drive and full-bridge drive so as to switch power modes under different communication distance requirements to achieve the aim of low power consumption. The amplified signal reaches a transformer and a matching network, the purpose is to increase the power conversion gain of a power amplifier and a transducer, the characteristic impedance of the transducer is measured by an impedance analyzer and then is matched, the tuning matching adopts a three-order Butterworth matching network, so that the equivalent load integrally tends to a pure impedance state, the reactive component in the power transmission process is reduced, the low power consumption is realized, the transformer has the impedance transformation function, the active resistance of the transducer can be transformed, the system meets the maximum power transmission, and the resistance value matching with the transducer is completed. And finally, transmitting the signals to an underwater acoustic channel through the transceiving combined underwater acoustic transducer for propagation.
Step four: preprocessing a received signal
The receiving and transmitting combined underwater acoustic transducer receives weak acoustic signals, the DSP judges the size of the received signals, the gain range of the automatic gain AGC chip is changed by sending control signals, so that input signals are between 0 and 5V, the receiving and transmitting times of different communication distances can be adaptively modulated to ensure normal and reliable awakening, the automatic gain control AGC chip selection chip AD605ARZ is a variable gain amplifier which has low noise and double channels and is linear in dB, and a single power supply with working voltage of +5V provides differential input and single-stage power control. The DSP then carries out synchronous judgment on the input signals, and the synchronous method comprises the following steps: the signal start position is found by performing a cross-correlation operation on data and m-sequence by using an overlap-save method (a known method), performing an overlap-save method of 17 consecutive windows in synchronization with a header, performing an overlap-save method of 3 consecutive windows in data, calculating a maximum peak value in each window, and finding a data start position corresponding to the maximum peak value.
Step five: demodulating a pre-processed received signal
And despreading and BPSK demodulation are carried out on the preprocessed received signal r (t) to obtain a demodulated signal r' (t).
The signals received by the receiver are selectively amplified and mixed to obtain the following parts: useful signal sI(t), channel noise nI(t), interference signal JI(t), i.e. the mixed signal is:
rI(t)＝sI(t)+nI(t)+JI(t)
the receiving end generates a pseudo-random sequence c '(t) which is the same as the transmitting end, and the received signal is multiplied by the local pseudo-random sequence c' (t) for despreading:
rI'(t)＝rI(t)c'(t)＝sI'(t)+nI'(t)+JI'(t)
wherein s isI'(t)＝sI(t)c'(t)＝a(t)c(t)c'(t)cosωc(t)＝a(t)cosωc(t)
Noise component nI' (t) is generally Gaussian band-limited white noise, the spectral density is basically unchanged but the relative bandwidth is changed after demodulation, the noise power is reduced, and an interference component JI' (t) is uncorrelated with the pseudo-random code, the spectral density is reduced and the interference power into the demodulation is only the same part of the signal band.
Step six: performing MELP decoding on the demodulated signal and playing
And (4) carrying out improved low-rate speech compression coding algorithm MELP decoding on the signals processed in the step five, synthesizing digital speech signals, and reducing the obtained digital speech signals into analog speech signals through a vocoder D/A converter and playing the analog speech signals through an earphone or a loudspeaker.
And unpacking the received code stream and sorting the parameters for the received voice bits. During decoding, the pitch period codeword containing unvoiced and voiced information is decoded first to determine whether the frame is unvoiced or voiced. The generation of the excitation signal is a key part of MELP decoding, and in one pitch period, the excitation signal is obtained through IDFT operation:
where T is a pitch period, which is the sum of the interpolated pitch value and the product of the pitch shake intensity and the pitch, and M (k) (1, 2.. 10) is an interpolated fourier series amplitude value, and M (T-k) (M (k)) is satisfied.
The speech synthesis is to pass the excitation signal through a synthesis filter, and then through gain correction, pulse discrete filtering, and post-filtering, finally obtain an analog speech signal y (t).
A block diagram of a physical structure of a low-power-consumption underwater digital voice communication system is shown in fig. 3, and the low-power-consumption underwater digital voice communication system comprises a power management module, a voice compression coding and decoding module, a signal processing and storing module, and a transmitting and receiving module. The power management module provides different power supply voltages for the voice compression coding and decoding module, the signal processing and storage module and the transmitting and receiving module as required, the voice compression coding and decoding module is connected with the signal processing and storage module through a serial port, and the signal processing and storage module is connected with the transmitting and receiving module through an A/D conversion module and a D/A conversion module. When the communication system is in a transmitting state, the input voice signal is transmitted after passing through the voice compression coding module, the signal processing and storing module and the transmitting module respectively; when the communication system is in a receiving state, the received signals are played after passing through the receiving module, the signal processing and storing module and the voice compression decoding module respectively.
The power supply management module introduces 18-28V direct current voltage from the outside to realize the management of input voltage;
the voice compression coding and decoding module comprises a microphone, a microphone output signal A/D acquisition module, a vocoder signal D/A converter and an earphone or a loudspeaker, when the system is in a transmitting state, after voice is input into the microphone, the output end of the microphone is connected with the input end of the microphone output signal A/D acquisition module, the output end of the microphone output signal A/D acquisition module is connected with the input end of the vocoder, and the output end of the vocoder is connected with the DSP input end of the signal processing and storage module; when the system is in a receiving state, the DSP output end of the signal processing and storing module is connected with the input end of the vocoder signal D/A converter, and the output end of the vocoder signal D/A converter is connected with an earphone or a loudspeaker.
The signal processing and storing module comprises a Digital Signal Processor (DSP), a serial port communication chip, a digital signal D/A converter, an analog signal A/D converter, a flash, a crystal oscillator circuit and the like, wherein the crystal oscillator circuit provides a clock for the DSP, and the flash is a program for testing a DSP storing system; when the system is in a transmitting state, the output end of a voice coder and decoder module vocoder is input into the DSP through a serial port communication chip MAX3223, the DSP outputs the processed voice to the input end of a digital signal D/A converter, and the output end of the digital signal D/A converter is connected with the input end of a power amplifier of a transmitting and receiving module; when the system is in a receiving state, the output end of the transmitting and receiving module band-pass filter is connected with the input end of the analog signal A/D converter, the output end of the analog signal A/D converter is connected with the input end of the DSP, and the output end of the DSP is connected with the input end of the voice coder-decoder module vocoder signal D/A converter;
the transmitting and receiving module comprises a power amplifier, a transformer, a matching network, an underwater acoustic transducer, a pre-stage amplifier, an automatic gain control AGC and a band-pass filter, when the system is in a transmitting state, the output end of a digital signal D/A converter of the signal processing and storing module is connected with the input end of the power amplifier, the output end of the power amplifier is connected with the input end of the transformer and the matching network, the output end of the transformer and the matching network is connected with the underwater acoustic transducer, and the underwater acoustic transducer transmits signals to an underwater acoustic channel; when the system is in a receiving state, the underwater acoustic transducer inputs a received signal into a triode for pre-amplification, the output end of the pre-amplification is connected with the input end of the automatic gain control AGC, the output end of the automatic gain control AGC is connected with the input end of the band-pass filter, and the output end of the band-pass filter is connected with the input end of the analog signal A/D converter of the signal processing and storage module.
FIG. 4 is a waveform diagram of a voice signal in a silent pool test, in which the waveform of the voice signal synthesized by the subjective test table name has a good fitting degree, high intelligibility, high definition and high naturalness.
Claims (7)
1. A low-power consumption underwater digital voice communication method is characterized by comprising the following steps:
the method comprises the following steps: MELP compression encoding of input speech signals
Inputting an analog voice signal x (t) into a microphone at a transmitting end by a vocoder, converting the analog voice signal x (t) through A/D, and performing low-rate compression coding on the converted digital voice through an improved low-rate voice compression coding algorithm MELP to obtain a compressed digital voice signal a (t);
step two: modulating compressed digital speech signals
Carrying out symbol mapping on the obtained compressed digital signal a (t), and preparing a modulation signal s (t) by adopting direct sequence spread spectrum and BPSK modulation;
the symbol rate of the compressed digital signal a (t) is RaC (t) is a pseudo-random sequence with a rate RcThe spreading process is a modulo-quadratic multiplication of a (t) and a pseudo-random sequence c (t), and the spread sequence d (t) is expressed as:
d(t)＝a(t)c(t)
moving d (t) to the carrier frequency for BPSK modulation, and the modulated signal s (t) is expressed as:
s(t)＝d(t)cosωct＝a(t)c(t)cosωc(t)；
step three: sampling the modulation signal to generate PWM wave and then transmitting;
sampling a modulation signal s (t) and a triangular wave t (n) to generate a PWM wave, generating a high-frequency triangular wave by utilizing DSP programming, simultaneously sampling the modulation signal and the triangular wave by using the same sampling rate, wherein the amplitude of the modulation signal is S (n), the amplitude of the triangular wave is T (n), comparing the amplitude of the modulation signal with the amplitude of the triangular wave, and outputting a high level if S (n) is more than or equal to T (n); if S (n) < T (n), outputting a low level;
the PWM wave generated by sampling is converted into sound wave through a D/A converter, a power amplifier, a transformer and a matching network, and finally the sound wave is converted into the sound wave through an underwater acoustic transducer to be transmitted in an underwater acoustic channel;
step four: preprocessing a received signal;
after the signal transmitted in the third step is transmitted through an underwater acoustic channel, the transducer receives and converts the transmitted weak acoustic signal into an electric signal, the electric signal is amplified in a front-mounted triode, AGC (automatic gain control) is carried out on an automatic gain control chip, a band-pass filter is carried out on the electric signal to obtain a bipolar analog signal, the analog signal is subjected to FFT (fast Fourier transform) conversion after A/D (analog to digital) conversion, and at the moment, the preprocessed received signal r (t) starts synchronous identification;
step five: demodulating a pre-processed received signal
Despreading and BPSK demodulation are carried out on the preprocessed received signals r (t) to obtain demodulation signals r' (t);
the signal received by the receiver is selectively amplified and mixed to obtain a useful signal sI(t) channel noise nI(t) and the interference signal, i.e. the mixed signal is:
rI(t)＝sI(t)+nI(t)+JI(t)
the receiving end generates a pseudo-random sequence c '(t) which is the same as the transmitting end, and the received signal is multiplied by the local pseudo-random sequence c' (t) for despreading:
rI'(t)＝rI(t)c'(t)＝sI'(t)+nI'(t)+JI'(t)
wherein s isI'(t)＝sI(t)c'(t)＝a(t)c(t)c'(t)cosωc(t)＝a(t)cosωc(t)；
Noise component nI' (t) is Gaussian band-limited white noise, and the spectral density is basically unchanged but the relative band is kept after demodulationWide variation, reduced noise power, interference component JI' (t) is uncorrelated with the pseudo-random code, the spectral density is reduced, and the interference power entering the demodulation is only the same part of the signal frequency band;
step six: performing MELP decoding on the demodulated signal and playing
Carrying out MELP decoding on the signal r' (t) processed in the step five by using an improved low-rate speech compression coding algorithm to synthesize a digital speech signal, reducing the digital speech signal into an analog speech signal y (t) by using a vocoder D/A converter, and playing the analog speech signal y (t) by using an earphone or a loudspeaker;
in one pitch period, the excitation signal is obtained by IDFT operation:
wherein T is a pitch period, which is a sum of an interpolated pitch value and a product of a pitch shake intensity and a pitch, M (k) is an interpolated fourier series amplitude value, k is 1, 2.. 10, and M (T-k) is M (k);
and (3) voice synthesis, namely, enabling the excitation signal to pass through a synthesis filter, and then carrying out gain correction, pulse discrete filtering and post-filtering treatment to finally obtain an analog voice signal y (t).
2. The low power consumption underwater digital voice communication method according to claim 1, characterized in that:
in the first step, the pitch period first defines a normalized autocorrelation function r (τ):
wherein the content of the first and second substances,
xk+m、xk+nrespectively representing the k + m and k + n samples of the analog voice signal x (t) after A/D conversion to generate digital signalSampling points, wherein m and n belong to (0, tau), the value range of tau is 40-160, and the final pitch period can be determined when the autocorrelation r (tau) value is more than or equal to 0.6;
and (3) the non-periodic voiced sound segment is determined by sub-band sound intensity extraction and quantization, and the peak value of the residual signal is calculated:
wherein r isnFor residual signals, if the peak a is greater than 1.34, the first sub-band is voiced, and if the peak is greater than 1.6, the first three sub-bands are voiced;
and quantizing according to the quantization codebook according to the generated pitch period and the determined non-periodic voiced segment to generate a compressed digital signal a (t).
3. The low power consumption underwater digital voice communication method according to claim 1, characterized in that:
in the fourth step, the step of synchronous identification is as follows: and performing cross-correlation operation on the data and the m sequence by adopting an overlap preservation method, adopting the overlap preservation method of 17 continuous windows when synchronizing the head, adopting the overlap preservation method of 3 continuous windows when synchronizing the data, calculating the maximum peak value in each window, and solving the data initial position corresponding to the maximum peak value to finish synchronous identification.
4. An underwater digital voice communication system with low power consumption using the method of claim 1, characterized in that:
the low-power-consumption underwater digital voice communication system comprises a power management module, a voice compression coding and decoding module, a signal processing and storing module and a transmitting and receiving module; the power management module provides different required power supply voltages for the voice compression coding and decoding module, the signal processing and storage module and the transmitting and receiving module, the voice compression coding and decoding module is connected with the signal processing and storage module through a serial port, the signal processing and storage module is connected with the transmitting and receiving module through an A/D conversion module and a D/A conversion module, and when the communication system is in a transmitting state, an input voice signal is transmitted after sequentially passing through the voice compression coding module, the signal processing and storage module and the transmitting module; when the communication system is in a receiving state, the received signals are played after passing through a receiving module, a signal processing and storing module and a voice compression decoding module respectively; the power management module introduces 18-28V direct current voltage from the outside to realize the management of input voltage.
5. The low power consumption underwater digital voice communication system according to claim 4, wherein:
the voice compression coding and decoding module comprises a microphone, a microphone output signal A/D acquisition circuit, a vocoder signal D/A converter, an earphone or a loudspeaker, when the system is in a transmitting state, after voice is input into the microphone, the output end of the microphone is connected with the input end for the A/D acquisition of the microphone output signal, the output end for the A/D acquisition of the microphone output signal is connected with the input end of the vocoder, and the output end of the vocoder is connected with the DSP input end of the signal processing and storage module; when the system is in a receiving state, the DSP output end of the signal processing and storing module is connected with the input end of the vocoder signal D/A converter, and the output end of the vocoder signal D/A converter is connected with an earphone or a loudspeaker.
6. The low power consumption underwater digital voice communication system according to claim 4, wherein:
the signal processing and storing module comprises a digital signal processor, a serial port communication chip, a digital signal D/A converter, an analog signal A/D converter, a flash and a crystal oscillator circuit, wherein the crystal oscillator circuit provides a clock for the DSP, and the flash is a program for testing a DSP storing system; when the system is in a transmitting state, the output end of a voice coder and decoder module vocoder is input into the DSP through a serial port communication chip MAX3223, the DSP outputs the processed voice to the input end of a digital signal D/A converter, and the output end of the digital signal D/A converter is connected with the input end of a power amplifier of a transmitting and receiving module; when the system is in a receiving state, the output end of the transmitting and receiving module band-pass filter is connected with the input end of the analog signal A/D converter, the output end of the analog signal A/D converter is connected with the input end of the DSP, and the output end of the DSP is connected with the input end of the voice coder-decoder module vocoder signal D/A converter.
7. The low power consumption underwater digital voice communication system according to claim 4, wherein:
the transmitting and receiving module comprises a power amplifier, a transformer, a matching network, an underwater acoustic transducer, a pre-stage amplifier, an automatic gain control AGC and a band-pass filter, when the system is in a transmitting state, the output end of a digital signal D/A converter of the signal processing and storing module is connected with the input end of the power amplifier, the output end of the power amplifier is connected with the input end of the transformer and the matching network, the output end of the transformer and the matching network is connected with the underwater acoustic transducer, and the underwater acoustic transducer transmits signals to an underwater acoustic channel; when the system is in a receiving state, the underwater acoustic transducer inputs a received signal into a triode for pre-amplification, the output end of the pre-amplification is connected with the input end of the automatic gain control AGC, the output end of the automatic gain control AGC is connected with the input end of the band-pass filter, and the output end of the band-pass filter is connected with the input end of the analog signal A/D converter of the signal processing and storage module.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202111346209.XA CN114157371A (en) | 2021-11-15 | 2021-11-15 | Low-power-consumption underwater digital voice communication method and communication system |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN202111346209.XA CN114157371A (en) | 2021-11-15 | 2021-11-15 | Low-power-consumption underwater digital voice communication method and communication system |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN114157371A true CN114157371A (en) | 2022-03-08 |
Family
ID=80459906
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN202111346209.XA Pending CN114157371A (en) | 2021-11-15 | 2021-11-15 | Low-power-consumption underwater digital voice communication method and communication system |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN114157371A (en) |
Citations (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6130859A (en) * | 1997-12-01 | 2000-10-10 | Divecom Ltd. | Method and apparatus for carrying out high data rate and voice underwater communication |
| US20060286931A1 (en) * | 2005-06-15 | 2006-12-21 | Mark Rhodes | Underwater communications system and method |
| CN202713317U (en) * | 2012-07-27 | 2013-01-30 | 中国科学院声学研究所 | Underwater modem |
| CN103310793A (en) * | 2013-05-16 | 2013-09-18 | 厦门大学 | Water sound real-time digital speech communication method |
| CN108880699A (en) * | 2018-07-06 | 2018-11-23 | 厦门大学 | A kind of underwater acoustic digital speed communication system and method under Quick moving platform |
| CN112291020A (en) * | 2020-10-15 | 2021-01-29 | 厦门大学 | Full-duplex underwater sound digital voice communication system and method thereof |
| WO2021109453A1 (en) * | 2019-12-02 | 2021-06-10 | 大连理工大学 | New high-robustness underwater optical communication system |
-
2021
- 2021-11-15 CN CN202111346209.XA patent/CN114157371A/en active Pending
Patent Citations (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6130859A (en) * | 1997-12-01 | 2000-10-10 | Divecom Ltd. | Method and apparatus for carrying out high data rate and voice underwater communication |
| US20060286931A1 (en) * | 2005-06-15 | 2006-12-21 | Mark Rhodes | Underwater communications system and method |
| CN202713317U (en) * | 2012-07-27 | 2013-01-30 | 中国科学院声学研究所 | Underwater modem |
| CN103310793A (en) * | 2013-05-16 | 2013-09-18 | 厦门大学 | Water sound real-time digital speech communication method |
| CN108880699A (en) * | 2018-07-06 | 2018-11-23 | 厦门大学 | A kind of underwater acoustic digital speed communication system and method under Quick moving platform |
| WO2021109453A1 (en) * | 2019-12-02 | 2021-06-10 | 大连理工大学 | New high-robustness underwater optical communication system |
| CN112291020A (en) * | 2020-10-15 | 2021-01-29 | 厦门大学 | Full-duplex underwater sound digital voice communication system and method thereof |
Non-Patent Citations (6)
| Title |
|---|
| TAE-HEEL WON、等: "Design and Implementation of an Omni-Directional Underwater Acoustic Micro-Modem Based on a Low-Power micro-Controller Unit", 《SENSOR 2012》 * |
| 孙卫华;王子勇;郭小宇;黄波;: "直接序列扩频水声通信技术研究", 舰船科学技术, no. 07, pages 2 * |
| 崔化超;赵安邦;周彬;孙坤平;崔岩;李桂娟;: "水下双工语音通信系统的设计与实现", 声学技术, no. 05 * |
| 张建利: "甚低速率语音编码算法研究", 《中国优秀硕士学位论文全文数据库》, pages 7 - 15 * |
| 王洪民;王晓阳;郭小宇;张琼;: "基于DSP的直接序列扩频水声通信技术研究", 舰船电子工程, no. 08 * |
| 郭中源;陈岩;贾宁;郭杰;陈庚;莫福源;马力;: "水下数字语音通信系统的研究和实现", 声学学报(中文版), no. 05 * |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN103971695B (en) | A kind of underwater digital voice communication system of channel self-adapting and its method | |
| US6119082A (en) | Speech coding system and method including harmonic generator having an adaptive phase off-setter | |
| CN103310793B (en) | A kind of water sound real-time digital speech communication method | |
| CN102576542A (en) | Determining an upperband signal from a narrowband signal | |
| US6138092A (en) | CELP speech synthesizer with epoch-adaptive harmonic generator for pitch harmonics below voicing cutoff frequency | |
| CN103714823B (en) | A kind of self adaptation subsurface communication method based on integrated voice coding | |
| JPH0713600A (en) | Vocoder ane method for encoding of drive synchronizing time | |
| CN105185382B (en) | A kind of imitative dolphin whistle underwater acoustic communication method to be stretched based on time-frequency spectrum | |
| CN108055087B (en) | Communication method and device for coding by using number of long-limb piloting whale sound harmonics | |
| CN102034480A (en) | Communication method of underwater digital voice | |
| US20090222264A1 (en) | Sub-band codec with native voice activity detection | |
| US7542813B2 (en) | Rapidly optimized wireless microphone system and method for controlling thereof | |
| EP2117140A1 (en) | A method of covertly transmitting information, a method of recapturing covertly transmitted information, a sonar transmitting unit, a sonar receiving unit and a computer program product for covertly transmitting information and a computer program product for recapturing covertly transmitted information | |
| CN109347568A (en) | A kind of polynary frequency modulation(PFM) underwater acoustic communication method of imitative dolphin whistle continuous phase | |
| CN102402986A (en) | Variable rate speech coding underwater acoustic digital speed communication method | |
| Woodward et al. | Digital underwater acoustic voice communications | |
| JPH0946233A (en) | Sound encoding method/device and sound decoding method/ device | |
| CN111976924A (en) | Real-time information communication device for diving full mask | |
| CN108880698A (en) | Communication means and device based on cetacean cry pulse time length | |
| CN205336318U (en) | Underwater sound speech communication terminal | |
| CN113395116A (en) | Underwater sound voice digital transmission method based on semantic compression | |
| CN114157371A (en) | Low-power-consumption underwater digital voice communication method and communication system | |
| CN113936680B (en) | Single-channel voice enhancement method based on multi-scale information perception convolutional neural network | |
| US5684926A (en) | MBE synthesizer for very low bit rate voice messaging systems | |
| Flanagan | Parametric representation of speech signals [dsp history] |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination |