82 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.

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 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.

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.

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 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 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.

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.

This disclosure provides systems, methods and apparatus, including computer programs encoded on computer storage media, for equalization domain selection at a wireless device. A wireless device may be configured to support a selection between time domain equalization and frequency domain equalization for receiving RF signals, which may be performed in accordance with detected channel conditions. For example, if a wireless device detects conditions associated with a relatively dispersive channel, the wireless device may select a frequency domain equalization and, if the wireless device detects conditions associated with a relatively flat channel, the wireless device may select a time domain equalization. In some implementations, performing a time domain equalization may include changing a sampling rate into a symbol rate, such as performing a resampling using a Farrow resampler. Such techniques may include a wireless device selecting an equalization scheme associated with a relatively lowest power consumption for detected channel conditions.

An optical transceiver includes an optical transmission unit and an optical reception unit and constitutes a network. The optical transmission unit transmits data at a symbol rate common in the network. The optical transceiver includes: an N-input-1-output optical switch that switches an optical signal received by the optical reception unit; a 1-input-N-output optical switch provided upstream of the optical signal with respect to the optical switch; and N delay lines of different lengths connected between the optical switch and the optical switch, and the optical signal input to the optical switch is input to the optical switch through one of the optical waveguides and received by the optical reception unit.

Method and system for automatic modulation classification (AMC) use time-series voltage signals representative of radio frequency (RF) signal envelope and frequency components as input features to a deep learning-based neural network, which enables classification of both modulation type and symbol rate without requiring in-phase and quadrature (IQ) demodulation. A feature extraction circuit captures RF signal envelope amplitude and frequency using stub-based sensing, and a Long Short-Term Memory (LSTM) neural network processes these features in a digitized form to classify the modulation and symbol rate with high accuracy and minimal latency.

The invention provides a frequency hopping system adaptive communication method based on bandwidth and service channel monitoring. The method comprises the following steps of: monitoring an interference state in an initialized communication bandwidth, and judging whether an available communication bandwidth in the initialized communication bandwidth is limited or not; if the available communication bandwidth is determined to be limited, calculating a limitation degree, and determining a corresponding adjustment parameter; calculating a target communication bandwidth and a target air interface symbol rate based on the adjustment parameter; pre-judging whether the transmission requirement of the service data with the highest priority is met or not under the target communication bandwidth and the target air interface symbol rate; if not, adjusting the time slot distribution of each piece of service data so as to meet the transmission requirement of the service data with the highest priority; synchronizing the target communication bandwidth and the target air interface symbol rate through the fixed TDMA time slot; and after the synchronization is completed, the nodes of the whole network switch the current communication bandwidth and the current air interface symbol rate to the target communication bandwidth and the target air interface symbol rate.

The invention discloses a multi-domain feature fusion communication signal modulation identification method based on deep learning, and belongs to the technical field of communication signal processing. The method comprises the following steps of: extracting a multi-domain feature vector from a complex baseband signal, wherein the multi-domain feature vector covers six categories including a time domain statistical feature, a frequency domain feature, a high-order cumulant feature, a differential phase feature, an I / Q component feature and a PSK (Phase Shift Keying) special feature; constructing a training data set by adopting a logarithmic space uniform sampling strategy, and realizing balanced coverage of a cross-order-of-magnitude symbol rate; and inputting the extracted feature vectors into a layer-by-layer dimensionality reduction neural network adopting a decline Dropout strategy for classification and identification. According to the method, the systematic coverage of 32 modulation types covering nine categories is realized, the natural immune characteristic of the high-order cumulant to additive white Gaussian noise is fully utilized, and the method has the advantages of wide identification range, strong anti-noise capability and good generalization performance.

This invention provides an adaptive variable rate method and apparatus based on signal-to-noise ratio (SNR) prediction, applicable to scattering communication. It addresses the adaptive variable rate requirement in scattering channels with fast fading characteristics, solving the problems of inaccurate SNR information and mapping relationships in determining the transmission symbol rate and coding / modulation scheme. The method includes: calculating an initial predicted SNR value and an SNR prediction error value based on an estimated SNR value, thereby obtaining an accurate predicted SNR value after error compensation; updating a dynamic threshold using the demodulation block error rate; obtaining an equivalent SNR using the predicted SNR value and the dynamic threshold; and determining the symbol rate and coding / modulation scheme by looking up a rate threshold table and a coding / modulation scheme threshold table using the equivalent SNR. This invention guides the transmission symbol rate and coding / modulation scheme with accurate SNR information and mapping relationships, effectively improving the average throughput of scattering communication systems and ensuring communication reliability.

The invention provides a single carrier signal sampling deviation estimation method and device. The method comprises the following steps: sampling a signal at an N-time symbol rate to obtain a sampling point sequence; selecting first 2NL sampling points in the sampling point sequence to calculate an initial timing error; calculating the offset of the current sampling point and the synchronization point according to the initial timing error, and updating the sampling point sequence; the timing error of each section is calculated in sequence; carmannl filtering is carried out on the timing error set, the slope of the timing error set is calculated through a sampling linear interpolation algorithm, and the sampling offset of the signal is calculated through the slope. According to the scheme, the whole signal is segmented, almost all data are used as samples for analysis, and the precision and correctness of an analysis result are ensured.

The invention provides a signal modulation converter and a signal conversion method for NRZ and PAM4, and belongs to the technical field of optical communication, and the signal modulation converter comprises a signal processing unit, an optical transmitting assembly and an optical receiving assembly. In the transmitting path, the signal processing unit converts a first electric signal which is received from the host interface and has a first modulation format into a second electric signal which has a second modulation format with a lower symbol rate; the light emitting assembly converts the second electric signal into a light signal and sends the light signal. In the receiving path, the optical receiving assembly converts the received optical signal into a third electric signal; and the signal processing unit converts the third electric signal into a fourth electric signal with the first modulation format and sends the fourth electric signal to the host interface. By reducing the symbol rate of the transmission signal on the optical fiber link, the dispersion loss of long-distance transmission is effectively overcome, and the transmission distance of the optical module can be remarkably increased.

The present application belongs to the technical field of wireless communication, and particularly relates to a multi-reflection device signal modulation and beamforming cooperation optimization method. The method of the present application mainly gives the modulation mode of the main system signal and the modulation order of the multi-reflection device, optimizes the signal constellation point of the reflection device and the beamforming to maximize the minimum Euclidean distance of the noise-free signal at the receiving end, thereby improving the symbol error rate performance of the main system and the Internet of Things communication. The scheme of the present application can be applied to the signal modulation of the main system and the Internet of Things system of any order, the scheme is simple to implement, and has strong application value.

The invention belongs to the technical field of signal generators, and particularly relates to an IQ baseband generation method and device at any symbol rate, and the method comprises the steps: obtaining IQ symbol data, carrying out the channel filtering of the IQ symbol data, and generating channel filtering IQ data with a predetermined bandwidth, the channel filtering IQ data having an original sampling rate; performing fractional resampling on the channel filtering IQ data to generate resampled IQ data, the fractional resampling being configured to increase an original sampling rate to a target sampling rate, the target sampling rate being defined as an integer fraction of a digital-to-analog conversion sampling rate; and carrying out integral multiple interpolation filtering on the resampled IQ data, increasing the target sampling rate to be consistent with the digital-to-analog conversion sampling rate, and carrying out digital-to-analog conversion to generate an IQ baseband signal. According to the method, fractional resampling is carried out on the channel filtering IQ data, so that the baseband generator can support continuous change and arbitrary setting of the symbol rate, and the universality and flexibility of the system are greatly improved.