117 results about "Symbol rate" patented technology

The invention discloses a time synchronization method suitable for a satellite communication system, and the method comprises the steps: analyzing a forward broadcast signal, and obtaining the reference time of a network clock; constructing a geometric prediction handle based on satellite orbit data, and predicting one-way propagation time delay between a terminal and a satellite by using the geometric prediction handle; estimating the processing delay of the terminal according to the symbol rate and the equipment working condition of the terminal and the delay calibration data; fusing the network clock reference time with the processing delay estimate to synthesize a terminal local time; and the sending time of the return link is calculated based on the local time and the propagation delay prediction. According to the method, the synchronization precision can be improved to a microsecond level, the convergence speed in a dynamic scene is accelerated, adaptive compression of the guard interval is supported, and the stability and spectrum efficiency of a satellite communication system are effectively improved.

The invention belongs to the technical field of signal processing, and particularly relates to a coherent light high-speed transmission signal processing method based on deep joint compensation, which comprises the following steps of: performing fine frequency offset and inter-carrier interference joint preprocessing on a coherent light receiving sampling sequence; obtaining a pre-processing output sequence and at least one estimation parameter derived from fine frequency offset and inter-carrier interference combined pre-processing; inputting the preprocessing output sequence into a diffusion type score guide joint compensation network module to obtain a deep joint compensation sequence; performing online judgment and decoding on the deep joint compensation sequence, and adaptively updating internal parameters of the diffusion type score guide joint compensation network module when a preset adaptive updating condition is met; according to the invention, the symbol residual error can be obviously reduced, the forward error correction gain can be improved, and the stability and robustness of the system under high-order modulation, high symbol rate and long-distance link scenes can be enhanced.

A test and measurement device has an input port to receive a signal from a device under test (DUT), the signal having a symbol rate, one or more analog-to-digital converters (ADC) to convert the signal to waveform samples at a sampling rate, and one or more processors, when aliasing is present: up-sample a portion of the signal having aliased samples to produce up-sampled samples; use the up-sampled samples to produce a real-time waveform; perform clock recovery on the real-time waveform to produce a recovered clock; and resample the aliased samples to produce a non-aliased waveform.

The invention provides a signal transmission method and device, and the method comprises the steps: receiving first information, the first information is used for indicating a first index corresponding to a first signal, the first signal is borne in a single carrier for transmission, the first index is used for indicating a first parameter group in a first incidence relation, and the first parameter group is used for indicating a second parameter group in a second incidence relation; the first association relationship comprises a plurality of indexes and a plurality of parameter groups corresponding to the plurality of indexes respectively, the first parameter group is a parameter group associated with the first index, the parameter group comprises a sampling frequency and / or a symbol rate, the sampling frequency is equal to a sampling frequency corresponding to a subcarrier interval of multiple carriers, and the symbol rate is equal to the sampling frequency corresponding to a subcarrier interval of the multiple carriers. The symbol period determined based on the symbol rate is equal to the symbol period of one subcarrier interval of the multicarrier, and transmitting the first signal according to the first parameter group, so that the communication device can transmit and receive a single-carrier signal and a multicarrier signal through the same set of parameters, thereby reducing the implementation complexity and saving the cost.

The invention provides a communication radiation source individual identification method based on time domain and time sequence coding constellation diagram feature fusion. The method is used for realizing stable identification of different transmitters under non-ideal conditions of carrier frequency offset, noise interference and the like. The method comprises the following steps: firstly, carrying out normalization and fixed-length interception on a received I / Q signal, and calculating an oversampling multiple according to a symbol rate and a time resolution; then, a time domain and time sequence coding constellation diagram double-branch feature extraction network is constructed, a time domain structure feature is extracted from a time domain branch by adopting a one-dimensional modified ResNet18, and a constellation diagram with time information is generated from a time sequence coding constellation diagram branch through a time sequence coding constellation module and the feature that the phase changes along with time is extracted; and carrying out joint modeling on the two types of features through a trainable weighted fusion mechanism. In the training stage, the performance of the model is improved by adopting cross entropy loss, an Adam optimizer, learning rate scheduling and an early stop strategy. In the reasoning process, input signals are classified by using the model weight with the highest verification accuracy, and a communication radiation source individual identification result is output. Experimental results show that the method can effectively enhance the frequency offset robustness and generalization ability of the model, realizes high-precision identification of the individual communication radiation source, and is suitable for the fields of security access of the Internet of Things, spectrum monitoring, electronic countermeasures and the like.

The invention relates to the technical field of signal analysis and processing, in particular to an FSK signal blind demodulation method and system in underwater acoustic communication signal analysis and processing, and the method comprises the steps: carrying out the coarse estimation of the bandwidth and carrier frequency of an underwater acoustic FSK signal; performing symbol rate estimation and timing synchronization based on an enhanced digital phase-locked loop; down-converting the received signal to zero intermediate frequency; carrying out discrete short-time Fourier transform to obtain a time-frequency domain distribution matrix of the signal; extracting time-frequency domain distribution in a time dimension by taking a symbol period as an interval, and taking out a frequency point with the strongest power at a corresponding moment to obtain a time-frequency distribution sequence of the signal; clustering the time-frequency distribution sequence by using K-means clustering, and judging the modulation order M of the FSK signal according to the contour value; and clustering the time-frequency distribution sequences into M classes, simultaneously obtaining the class to which the frequency corresponding to each sampling point in the sequences belongs, and judging the class as an M-system symbol until the blind demodulation is finished. According to the invention, blind demodulation of FSK signals with guard intervals and without guard intervals can be realized.

The invention discloses a resource allocation method and device, equipment, a storage medium and a product, and relates to the technical field of communication, and the method comprises the steps: obtaining a symbol rate and a historical demand bandwidth supported by a satellite terminal in response to a received login request of the satellite terminal; selecting or determining a candidate sub-band set from pre-divided sub-bands based on the symbol rate supported by the satellite terminal and the historical required bandwidth; respectively calculating discreteness indexes of corresponding spectrum utilization rates when the satellite terminal accesses each sub-band in the candidate sub-band set; and selecting or determining a target sub-band from the candidate sub-band set based on the discreteness index of the spectrum utilization rate, and allocating the target sub-band to the satellite terminal. According to the invention, the sub-band optimization is carried out according to the discreteness index of the corresponding spectrum utilization rate when the satellite terminal accesses each available sub-band, and the balanced distribution of base station resources is realized, so that the spectrum utilization rate of the base station is improved, the burden of the base station is reduced, and the use cost is reduced.

The invention provides a communication perception fusion waveform design method based on space-time constructive interference, which belongs to the field of wireless communication and comprises the following steps: establishing a downlink multi-user communication perception fusion system model; establishing a communication model and a sensing model; establishing an optimization problem for minimizing the transmitting power, wherein the optimization problem has a space-time constructive interference constraint condition and a CRB constraint; after the CRB constraint of the optimization problem is relaxed into linear constraint by using a continuous convex approximation technology, a second optimization problem is obtained through iterative solution, the second optimization problem is simplified through a Lagrangian multiplier method and a Carush-Kuhn-Tuck condition, and the simplified second optimization problem is solved through iterative solution of a main optimal minimization algorithm; and obtaining a designed transmitted waveform vector. According to the invention, the symbol error rate of transmission and the transmitting power of the transmitting end can be effectively reduced under the condition that the communication and sensing performance requirements are met.

This invention discloses a radar embedded communication method based on OTFS modulation. The implementation steps are: 1. Generating an integrated signal based on OTFS modulation; 2. Estimating the relative speed and distance of the RF tag by the integrated signal receiver; 3. Modeling the time-domain channel based on the speed and distance measurement results of the RF tag by the integrated signal receiver; 4. Equalizing the received signal using the constructed time-domain channel matrix and performing decorrelation decision on the equalized signal. This invention utilizes the characteristic that OTFS signals can obtain full channel diversity in both time and frequency, achieving a more accurate estimate of the time-varying frequency-selective bidispersive wireless channel, and equalizing the fading signal, reducing the communication symbol error rate and effectively improving the communication reliability of the radar embedded communication system.

The invention discloses an anti-high dynamic Doppler low signal-to-noise ratio satellite signal blind detection method, which comprises the following steps of: performing frequency domain analysis on a delay correlation signal according to a signal symbol rate, and dividing a signal band and a noise band near the delay correlation signal to construct a detection quantity to perform constant false alarm rate (CFAR) detection; and finally, obtaining a final signal existence or disappearance result through time consistency judgment. According to the method disclosed by the invention, the analytic CFAR threshold and the state machine smoothing are combined, and the problem of detection failure in a low signal-to-noise ratio and high dynamic Doppler environment in the prior art is solved.

The invention discloses an encryption burst communication receiving method based on an FPGA, and belongs to the technical field of burst communication. According to the invention, the problems of poor synchronization performance and poor anti-interference capability at the receiving end in the existing method are solved. According to the method, the carrier synchronization range is expanded by introducing the pilot frequency to enable the synchronization heads to be '1', the limitation that a conventional frequency offset estimation algorithm can only work under the condition that the Doppler frequency shift is smaller than 0.5 times of the symbol rate is broken through, and the capacity of the system for adapting to the high-dynamic environment is improved. A spreading code continuous tracking working mode is designed, coarse estimation and synchronization are carried out in a frame header part, spreading codes are still adopted in an information part, and continuous tracking is carried out, so that the system can adjust the phase of the spreading codes in real time in the whole frame to solve the problem of phase error accumulation caused by adopting a low-precision clock; the problems that when one frame of data at the sending end is long, the synchronization performance is reduced when a low-precision clock is adopted, and even effective synchronization cannot be achieved are solved. The method can be applied to the field of burst communication.

Methods, systems, and devices for a wakeup receiver operation is described. A receiver may include a conversion circuit that converts an RF signal to a baseband signal, where the baseband signal comprise a set of symbols received at a symbol rate. The receiver may include an analog-to-digital converter that converts the baseband signal to samples at a sample rate greater than the symbol rate. The receiver may include a set of correlators, each correlator of the set may receive a respective subset of the samples of the baseband signal and generate a respective output. The receiver may include a compare circuit that receives the respective outputs from the set of correlators and compares the respective outputs with a threshold, where the compare circuit also generates a wakeup command based at least in part on at least one output of the respective outputs satisfying the threshold.

The invention discloses a front-feedback combined burst signal fast symbol synchronization method, which relates to the technical field of digital signal processing and satellite communication, and comprises the following steps: firstly, sampling a received signal subjected to matched filtering processing according to a 4-time symbol rate by using an initial NCO (Network Controlled Oscillator), then dividing a sampling value into two paths, sending one path to an FIFO (First In First Out) register for caching, and using Oamp in the other path; the method comprises the following steps of: firstly, performing timing deviation estimation by using an FIFO (First In First Out) register and an M algorithm, secondly, determining a moment when the FIFO register starts to read data, an oscillation initial value of an interpolation control NCO and an initial value of an interpolation coefficient according to a timing deviation estimation result, starting the interpolation control NCO, and then, sending the data read by the FIFO register into a Farrow structure interpolation filter for interpolation processing according to the initial value of the interpolation coefficient. And finally, a Gardner loop is started, feedback control is carried out on the interpolation control NCO, and an interpolation result is output, so that the fast symbol synchronization speed and precision of the burst signal are improved, and the stability and timeliness of fast symbol synchronization of the burst signal are guaranteed.

Improved techniques for estimating parameters of a jamming signal. An input signal is identified. This input signal is suspected of being a jammed composite signal. Attributes of a reference signal are determined. The reference signal is an expected signal that was expected to be received. A form fitting operation is performed in which the reference signal is formed fitted with the input signal. The reference signal is subtracted from the input signal to generate an isolated output signal. A suspected portion of the isolated output signal is identified. An estimated symbol rate and an estimated center frequency for the jamming signal are determined based on the suspected portion. The estimated symbol rate and the estimated center frequency are used to facilitate a subsequent mitigation operation of eliminating or reducing an impact of the jamming signal against the signal of interest.

The invention discloses a signal restoration method, device and system. The method comprises the following steps: receiving a signal processing request; in response to the signal processing request, segmenting the composite signal to obtain a plurality of signal blocks; converting to obtain a plurality of frequency domain signal blocks in a frequency domain; according to the symbol rates corresponding to the multiple signal types, determining signal interception numbers corresponding to the multiple signal types; according to a time sequence corresponding to the plurality of frequency domain signal blocks, sequentially carrying out interception processing on the plurality of frequency domain signal blocks to obtain frequency domain sub-signals respectively corresponding to the plurality of signal types until frequency domain total signals respectively corresponding to the plurality of signal types are obtained; converting to obtain a plurality of time domain total signals; and according to the time domain total signals corresponding to the multiple signal types, target restoration signals corresponding to the multiple signal types are determined. According to the method and the device, the technical problems that signal separation is difficult to accurately carry out and signal restoration accuracy exists due to the fact that signal interception parameters are difficult to flexibly configure in related technologies are solved.

The invention relates to a method for frame synchronization for a digital receive signal, which is received at an output of a transmission channel as a consequence of a binary digital transmit signal supplied to an input of the transmission channel, wherein the binary digital transmit signal contains information to be transmitted in a data packet which has a predetermined number D of data symbols in a data section and a predetermined number S of synchronization symbols in an optional synchronization section, and wherein the binary digital transmit signal has a predetermined symbol rate, wherein the digital receive signal is represented by a sequence of digital samples and wherein a sampling frequency is used which is greater than or equal to the symbol rate.According to the invention, a sliding window is defined which has a temporal length corresponding to the temporal length of a data packet or a data segment of a data packet. At each position µ of the sliding window, a sample value is assigned to each symbol segment of the data segment and optionally to each symbol segment of the synchronization segment of the sliding window. For each position µ of the sliding window, a position value Λ is determined using a predefined metric. met[µ] is determined, using at least the sampled values ​​assigned to the symbol segments of the data segment. The metric is based on the use of a model for the transmission channel, which assumes a memory-free transmission channel and in which the digital received signal is described by a conditional probability density function for each of the possible binary values ​​of the transmitted signal. From the initial time of the sliding window in a position for which the position values ​​Λ met assume a maximum or for which the position value Λ met If a predetermined threshold is exceeded, a start time of a data packet in the digital received signal is determined. The invention further relates to a device and a computer program for implementing the method.

A receiver includes an optical front-end and digital circuitry. The optical front-end is configured to receive an optical signal including first and second optical signal components having first and second polarizations and modulated with symbols at a symbol rate. The digital circuitry is configured to derive first and second digital signals representing the first and second optical signal components having the first and second polarizations. The digital circuitry includes a clock detector configured to calculate correlation terms, the correlation terms being calculated in a frequency-domain with a frequency offset commensurate with the symbol rate. The clock detector is configured to recover a clock signal of the symbols by (i) summing selected pairs of the correlation terms, and (ii) calculating or estimating a sum-of-squares of the summed pairs.

Provided are a signal transmission method and device, the method comprising: receiving first information, the first information being used for indicating a first index corresponding to a first signal, the first index being used for indicating a first parameter group in a first association relationship, the first association relationship comprising a plurality of indexes and a plurality of parameter groups corresponding to the plurality of indexes, the first parameter set is a parameter set associated with the first index, the parameter set comprises a symbol rate, a first parameter and a cyclic prefix length, the first parameter is a parameter of a filter used for transmitting the first signal, and the cyclic prefix length is the length of a cyclic prefix contained in each symbol in the first signal; and the first signal is transmitted according to the first parameter group, so that interference of different degrees on the signal caused by the waveform of the filter is relieved, and the performance stability of the receiver is improved.

The application provides a blind demodulation analysis method of a signal, a chip, a remote control device and a storage medium. The method comprises the following steps: determining the center frequency of a target signal through a bandwidth scanning strategy; performing mixing processing and analog-to-digital conversion on the target signal to obtain an I-path intermediate frequency digital signal and a Q-path intermediate frequency digital signal, so as to determine the amplitude of the target signal; determining the modulation mode of the target signal according to the change of the amplitude of the target signal within a preset time length; if it is determined that the modulation mode is OOK, determining the amplitude mean value of the target signal, and demodulating the target signal according to the size relationship between the amplitude of the target signal and the amplitude mean value to obtain a baseband signal; if it is determined that the modulation mode is 2FSK, determining the frequency offset estimation mean value and the frequency offset of the target signal, and demodulating the target signal according to the size relationship among the frequency of the target signal, the center frequency of the target signal and the frequency offset estimation mean value to obtain the baseband signal; and performing frequency measurement on the baseband signal to determine the symbol rate.

The present application relates to the technical field of communications. Disclosed are a data transmission method and a communication apparatus, which can reduce a symbol error rate in a data transmission process when signal dependent noise is considered to be used for designing a sending filter sequence. The method comprises: receiving first information, wherein the first information is used for indicating at least one of the intensity of signal dependent noise measured by a receiving end, and the ratio of the intensity of the signal dependent noise measured by the receiving end to the intensity of signal independent noise; and sending first data, wherein the first data is data formed by means of shaping a sending pulse forming waveform by a sending end, and the sending pulse forming waveform is determined on the basis of the first information. The embodiments of the present application are used for taking into consideration the impact of signal dependent noise on a sending filter sequence and a receiving filter sequence.

The invention relates to a method for frame synchronization for a digital receive signal, which is received at an output of a transmission channel as a consequence of a binary digital transmit signal supplied to an input of the transmission channel, wherein the binary digital transmit signal contains information to be transmitted in a data packet which has a predetermined number D of data symbols in a data section and a predetermined number S of synchronization symbols in an optional synchronization section, and wherein the binary digital transmit signal has a predetermined symbol rate, wherein the digital receive signal is represented by a sequence of digital samples and wherein a sampling frequency is used which is greater than or equal to the symbol rate.According to the invention, a sliding window is defined which has a temporal length corresponding to the temporal length of a data packet or a data segment of a data packet. At each position µ of the sliding window, a sample value is assigned to each symbol segment of the data segment and optionally to each symbol segment of the synchronization segment of the sliding window. For each position µ of the sliding window, a position value Λmet[µ] is determined using a predefined metric, whereby at least the sample values ​​assigned to the symbol segments of the data segment are used. The metric is based on the use of a model for the transmission channel, which assumes a memory-free transmission channel and in which the digital received signal is described by a conditional probability density function for each of the possible binary values ​​of the transmitted signal.From the initial time of the sliding window in a position for which the position values ​​Λmet reach a maximum or for which the position value Λmet exceeds a predetermined threshold, a start time of a data packet in the digital received signal is determined. The invention further relates to a device and a computer program for implementing the method.

The application discloses a signal detection and identification method, comprising the following steps: S1, determining the specific specification of the signal to be detected and identified; S2, extracting the signal characteristics of the signal with the specific specification; S3, distinguishing the signal bandwidth and signal type on the frequency point based on the signal characteristics; S4, locating the accurate burst starting position based on the instantaneous phase curve of the signal or the unique code information of the signal; accurately determining the end position of each burst based on signal demodulation or the frame structure of the signal; S5, counting the repetitive characteristics of the burst starting position, and calculating the frequency offset and / or signal-to-noise ratio information of the signal based on the repetitive characteristics. The corresponding system, electronic equipment and computer readable storage medium are also disclosed, which can detect the target digital signal under the condition of noise packets and other interference, and the accuracy reaches within one symbol error of 98% or above, and the modulation, symbol rate and related information characteristics of the target signal are extracted.

The invention discloses an automatic detection method for an uplink of a ground measurement and control station, and relates to the technical field of aerospace measurement and control and satellite communication operation guarantee. The separability of the uplink signal in the aliasing environment is obviously improved; candidate components are obtained based on blind source separation with mutual information minimization as a target, modulation identification is carried out through a depth model driven by time-frequency representation, and robust distinguishing of different satellite service systems is achieved. The carrier frequency, the symbol rate, the modulation category and the task parameter set are subjected to weighted similarity matching in a tolerance range, and one-to-one constraint solution is carried out, so that mismatching and misjudgment caused by near-frequency, near-speed or fast switching can be effectively inhibited; when observation environment or orbit parameters change, separation and window configuration can be adaptively adjusted, and real-time performance and accuracy are maintained.

The invention relates to the technical field of data processing, and particularly provides a method, a system, equipment and a medium for restoring an unknown signal constellation diagram, and the method comprises the steps: carrying out the self-correlation calculation of an in-phase I-path component of a received signal, carrying out the spectrum analysis of a self-correlation result, so as to enhance a spectrum component corresponding to a symbol rate, estimating the code element rate of the signal from the frequency spectrum; based on the code element rate, analyzing energy change characteristics of I-path and orthogonal Q-path components of the received signal, and detecting and identifying a code element boundary position to obtain a code element boundary sequence; and for each code element interval divided by the code element boundary sequence, aggregating I-path and Q-path sampling values in a stable region of each code element interval to obtain constellation point coordinates corresponding to each code element, and generating a constellation diagram by all constellation points. According to the invention, clear and accurate constellation diagram recovery of various modulation signals, especially high-order QAM signals, under the condition of low signal-to-noise ratio is realized.

The present invention discloses a QPSK baseband signal symbol rate estimation method based on constellation diagram variance, the steps of which are as follows: obtaining baseband signal sequences of different lengths; obtaining constellation diagrams corresponding to baseband signal sequences of different lengths; calculating the mean of constellation points in each quadrant of the constellation diagram; obtaining a mean residual diagram of each constellation diagram; calculating the total variance of each mean residual diagram; determining a sampling invariant point; and estimating the symbol rate of the QPSK baseband signal. The present invention uses the total variance of the mean residual diagram to determine the sampling invariant point, and the number of total variances of the calculated mean residual diagram is the number of sampling points in one symbol period. The sampling invariant point is used to estimate the symbol rate of the QPSK baseband signal, thereby avoiding the influence of the band-limited filter and being applicable to the symbol rate estimation of the baseband signal using a band-limited shaping filter. The present invention has the advantages of high flexibility, low computational complexity, and high engineering application value.

The invention relates to the technical field of wireless communication, and discloses a double sampling and bandwidth optimization data processing method for a low-rate FSK (Frequency Shift Keying) signal, which comprises the following steps: receiving and processing an FSK radio frequency analog signal through a radio frequency receiving front-end module, setting a multiple symbol rate through a double sampling demodulation module, receiving and amplifying the FSK radio frequency analog signal through the multiple symbol rate, and outputting the amplified FSK radio frequency analog signal through a bandwidth optimization demodulation module. And the digital signal processing module processes the double sampling demodulation data by using an algorithm to restore the initial data. According to the invention, a high-density time domain sequence is generated by adopting a double sampling rate, so that the bandwidth of a digital filter of a receiver is increased, normal communication can still be maintained when frequency deviation exists between a transmitter and the receiver, and even if the FSK receiver has a high error rate problem caused by the frequency deviation problem at a low rate, the communication efficiency is improved. And low-rate and long-distance communication can still be realized.

The application discloses a kind of communication signal parameter blind estimation methods, devices and program products for intelligent instrument, involve signal identification field, to with low complexity calculation cost, accurately estimate the carrier frequency offset value of electromagnetic signal, symbol rate and pulse shaping filter type.The application carries out normalization to received signal, carrier frequency offset rough correction, estimates carrier frequency offset and meets rate, carries out the pre-estimation of pulse shaping filter type from the spectrum diagram of rough correction signal, and, for square root raised cosine filter / Raised cosine filter, first sequentially carrier frequency offset correction is carried out to first signal, based on square root raised cosine filter Matching filter, and based on symbol rate construction constellation diagram, finally according to constellation diagram, distinguish whether square root raised cosine filter or Raised cosine filter.The application realizes comprehensive, accurate, reliable estimation to electromagnetic signal parameter with low calculation complexity, realizes the automation and wisdom of electromagnetic signal measurement.

The invention relates to a software radio burst waveform-based synchronization method, a software radio burst waveform-based synchronization system, software radio burst waveform-based synchronization equipment and a medium in the technical field of electronic communication. A configured one-way sliding window synchronization algorithm is adopted at a receiving end to carry out preamble capture on a burst signal in a burst communication scene based on software radio, and a system can automatically select a proper sliding window size according to the designed one-way sliding window synchronization algorithm, so that a preamble sequence of a burst communication frame is correctly captured and a baseband symbol rate is adapted; according to the invention, the receiving end can correctly decode the information data, the correlation between the receiving sequence and the leader sequence is ensured, the hardware calculation overhead is reduced, the system is simple to implement, the hardware requirement is low, and the practicability is high.

This invention proposes a low-complexity time-domain parallel clock recovery method based on a 4 / 3 sampling rate. First, a new error extraction model is proposed. Interpolation processing based on a sinusoidal interpolation algorithm is introduced to achieve clock offset compensation. Next, loop filtering and NCO parallel processing are designed specifically for 4 / 3 symbol rate sampling. After interpolation, redundant sampling points are removed through decision making and elastic buffering. Finally, a sinusoidal interpolation algorithm with fixed interpolation coefficients is used to recover the optimal sampling points under 4 / 3 symbol rate sampling, achieving reliable and accurate recovery of transmitted information. This method can achieve high-precision and high-stability parallel time-domain clock recovery in a 4 / 3 symbol rate sampling receiving system, making it particularly suitable for resource-constrained and limited satellite communication systems.

The application discloses a kind of soft demodulation methods for low signal-to-noise ratio 188-110A signal under large frequency offset, it is related to communication technical field, comprising: S1, preamble sequence detection and preamble sequence coarse synchronization based on autocorrelation;S2, prepare baseband preamble reference modulation signal;S3, preamble sequence fine synchronization based on differential cumulative correlation;S4, M&M and L&R joint frequency offset estimation and frequency offset correction based on preamble reference signal;S5, according to D1, D2 value table in 188-110A protocol, the demodulation of interleaving indicating D1, D2;S6, the frame search and frame demodulation of data are executed;S7, after the frame constellation of data is disturbed, the soft information of data frame is obtained;S8, based on 188-110A protocol, the deinterleaving of data frame soft information is completed by corresponding bit storage and reading mode, then it is decoded.After compared with prior art, signal detection anti-frequency offset is improved from ±0.4% to ±100%, signal detection performance is improved about 3dB gain, signal demodulation decoding performance is improved about 2.6dB gain.