61 results about "Dsp algorithms" patented technology

Audio transducers (headphones, speakers, microphones) inherently do not accurately reproduce the signal presented to them at the input. This can be compensated for by taking into account the transducer characteristics and transforming the input signal using a digital signal processor (DSP) to counteract the inaccuracies. However, for the compensation to take place, the DSP needs to know the characteristics of the transducer. For systems with built-in transducers (like laptops with internal speakers) the characteristics of the internal speakers can be stored on the hard-drive of the laptop and the DSP can read this data and make the appropriate compensations. Because a transducer (headphone, speaker, microphone) has its own characteristics that need to be compensated for separately, a profile is supplied to the DSP either by a database lookup based on an identification made by the user or transducer itself or by profile information stored on the transducer. Once the characteristics of a transducer are known, many additional DSP algorithms can be applied in order to improve the audio performance and even safety of the system.

The invention discloses a nonlinear phase noise compensation method and a system in a coherent optical fiber communication system. Signals enter the optical fiber from the transmitting end through themultiplexer and are transmitted to the receiving end. After coherent receiving and ADC sampling, the signals enter a receiving end DSP, wherein the receiving end DSP comprises IQ orthogonalization, dispersion nonlinear compensation, sampling clock recovery, self-adaptive channel equalization, carrier frequency offset estimation, carrier phase estimation and self-adaptive nonlinear phase tracking;initializing parameters of the adaptive nonlinear phase tracking; judging the input signal to generate a reference signal; calculating a gain factor corresponding to the sampling point according to the reference signal; updating the weight coefficient and the autocorrelation matrix of the input signal according to the gain factor so as to generate an output signal; repeating the above steps, andobtaining the output of the nth iteration through the data recursion of the (n-1) th iteration. The method does not need extra optical devices, is suitable for any system structure and modulation mode, and the DSP algorithm of the receiving end is low in calculation complexity.

Interconnection bus including longitudinal interconnection bus and transverse interconnection bus interconnected through reconfigurable switch network are setup between reconfigurable processing units arranged in array. Reconfigurable processing units are connected to each other longitudinally through basic unit data lines, which are connected to transverse interconnection bus through reconfigurable switch network. In transverse adjacent reconfigurable processing units at same stage of pipeline, their adders, shifters, accumulators, and register are connected through interconnection lines containing control switch. Reconfigurable processing unit itself of using COROIC algorithm possesses high reconfigurability, providing features of realizing wide used DSP algorithms, simple structure and rules, easy of modularized. Thus, the reconfigurable processing unit is suitable for being as core unit in reconfigurable chip.

The invention relates to an MIMO channel simulator control method and the implementing device thereof, relating to wireless communication simulating system. And it comprises DSP algorithm module, synchronous dynamic memory, flash memory, asynchronous serial communication module and level converting module, and considers both time and frequency fading and selective space fading, and considers corresponding selective frequency fading by PDP, corresponding selective time fading by DPSD and corresponding selective space fading by PAS in the model establishing course. And the simulator has low price and simple structure.

The invention discloses a data processing method and a data processing device. The data processing method comprises the steps of dividing data received by a receiver into first path data and second path data, acquiring burst packet starting point information and a transient frequency offset estimation value of the first path data, determining starting point information of the second path data according to the burst packet starting point information, carrying out transient frequency offset compensation on the second path data according to the starting point information of the data and the transient frequency offset estimation value, carrying out dispersion equalization processing on the second path after transient frequency offset compensation, and carrying out a subsequent data recovery operation on the second path data after dispersion equalization processing. The data processing method disclosed by the invention solves a problem of power frequency offset transient effects brought about by laser switching in related technologies, eliminates mutual effects among different DSP algorithms, compensates great line width and the frequency offset transient effects brought about by laser switching, and improves the channel utilization ratio of an optical burst coherent reception system.

The invention provides a method for a microwave photon link based on a full light spectrum utilization and post-compensation technology, which comprises the following steps: loading a received microwave signal onto light by using two cascaded electro-optical phase modulators and carrying out all-optical down frequency conversion; depending on a mould composed of a polarization-maintaining circulator, a fiber Bragg grating (FBG) and a balanced detector, simultaneously utilizing a transmission spectrum and a reflection spectrum of a grating; and then utilizing cosine and sine functions for DSP (Digital Signal Processing). The method has the beneficial effects that the system gain is effectively improved by utilizing the energy of a whole light spectrum, and the system linearity is improved by utilizing a DSP algorithm with lower complexity, and further, the dynamic range is improved.

A quasi adaptive acoustic equalization method is characterized by: using an adaptive digital filter and a corresponding DSP algorithm module to realize changing of a sound source signal and carry out adaptive processing to a whole audio frequency pass band; using a quasi adaptive acoustic overall equalizer, which combines modal equalization, phase equalization and amplitude equalization, to process the whole audio frequency pass band; firstly carrying out low band modal equalization to a low pass channel of a frequency division network; then superposing high and low pass sound signals in an area which is close to a frequency division point on a high pass channel and carrying out the phase equalization; and then carrying out the amplitude equalization to a whole frequency band and a high frequency band so as to realize comprehensive optimization processing to the whole audio frequency pass band. The general equalization means that a measurement microphone and several DSP algorithm modules form a modal equalizer module, a phase equalizer module and an amplitude equalizer module respectively so as to realize and have effect on a loudspeaker system. The three modules can be realized by using a digital filter with an adjustable parameter. Finally, the needed audio signal can be obtained.

A software implementation of a digital signal processing function is protected by selecting a subset of parameters (210) of the signal processing function and embedding a watermark (230) in the selected parameters.

A method of testing the performance of real time DSP algorithms after customer code has been added and includes the steps of embedding a signature equation in the DSP code that calculates to a given signature value at the output of the DSP when running a test mode program through the DSP.

A method and circuit arrangement utilize a register file of an execution unit as a local instruction loop buffer to enable suitable algorithms, such as DSP algorithms, to be fetched and executed directly within the execution unit, and often enabling other logic circuits utilized for other, general purpose workloads to either be powered down or freed up to handle other workloads.

The invention relates to an NGEO satellite routing algorithm based on fuzzy theory, characterized by comprising: employing a fuzzy theory method to calculate an FSCI according to satellite node information; calculating the determine index of a routing algorithm, and finding a DSP according to a DSP routing algorithm; in case of satellite congestion, a satellite notifying adjacent satellites of FSCI signaling information; after the satellites receiving the FSCI signaling information, detouring according to the detouring strategy of an improved ELB algorithm and priorities of different packages. The NGEO satellite routing algorithm has a reasonable design, uses the fuzzy theory to obtain the FSCI and the detouring strategy, is simple to compute, meanwhile well distributes flow, and realizes satellite constellation load balancing. Compared with a traditional DSP (Dijkstra DSP algorithm) and an ELB (displaying load balancing) routing algorithm, the NGEO satellite routing algorithm has remarkably improved performance indexes.

The invention relates to a channel equalization method based on SHAP feature optimization, and the method comprises the steps: (1) enabling a transmitting end to transmit data containing a training sequence to a receiving end through an optical fiber, and enabling the receiving end to input the sampled data containing the training sequence to an equalizer module based on machine learning; (2) extracting a received training sequence by an equalizer module based on machine learning, and constructing a feature vector for the training sequence; (3) obtaining the SHAP value of each feature of each sample through calculation, and measuring the feature weight importance of each feature and the influence on the prediction result based on the index; (4) selecting features based on the feature weights, and constructing feature vectors for the training set and the test set according to a feature selection result; and (5) inputting the constructed feature vector into an equalizer based on a DSP algorithm to equalize subsequent effective data. According to the invention, the system performance is maintained, the influence of clock jitter on the system performance is reduced, and the calculation complexity of channel equalization is reduced.

The invention discloses a coherent light OFDM system receiving end high-speed data parallel processing structure and method. The structure comprises a DC coarse equalization module, a parallel symbol synchronization module, a parallel FFT window module, a subsequent DSP processing module and a data recovery module. The method comprises the steps that firstly the characteristics of dispersion compensation are utilized so that a high-speed serial data flow string is converted into a parallel data flow string and then reduced to be within the processable range of a chip clock; then double training symbols are arranged and a double related peak decision mechanism is introduced so that symbol synchronization parallel processing is achieved and fractional frequency offset estimation and compensation functions are completed; and finally partitioning processing is performed on multipath parallel data. The requirement of a high-speed optical fiber communication system for ADC/DAC sampling rate and DSP algorithm chip processing speed is reduced by the structure and the method so that a problem that a DSP algorithm processing chip cannot meet the real-time processing requirement of the system is solved.

The invention discloses a microwave photon channelization receiving method based on polarization multiplexing optical frequency dispersion and an integrated coherent receiver, and relates to the technical field of microwaves and the technical field of optical communication. The method is as shown in Figure 1 in the specification, and a light source, a radio frequency signal source, a Mach-Zehndermodulator, an optical coupler, an erbium-doped fiber amplifier, an optical band-pass filter, a double-parallel quadrature phase shift keying modulator, a polarization controller and an integrated coherent receiver (ICR) are included. And channelized receiving of four channels can be realized only by using two linearly polarized multiplexed local oscillator optical frequency combs. In addition, according to the method, the problem of mirror image interference existing in the dual-optical-frequency microwave photon channelization receiver is solved by combining ICR with a digital signal processing DSP algorithm. Because no high-frequency electric device is used, the method has the advantages of large bandwidth, high image rejection ratio and the like. Due to the fact that optical carriers are restrained and balance detection is adopted, the method further has the advantage of being high in gain.

In the case of a method for optimizing the transfer behavior of loudspeaker systems in a consumer electronics device, the actual transfer function of the loudspeaker system to be optimized is determined, in that a test signal is reproduced on the loudspeaker system either directly or via audio signal processing stages (DSP, amplifier, etc.) installed in the device, and the acoustic signal emitted by the loudspeaker system is captured by means of microphone and the actual transfer function is determined from the measurement values. Furthermore, in the method, a code for a DSP algorithm is generated and optimized to the extent that, in the case of a previously set maximum deviation from the desired transfer function, the fewest DSP resources are needed. The completely optimized code is loaded into the DSP of the device and activated. A corresponding arrangement has a microphone, a measurement unit, a DSP code generator, and an interface that loads the generated DSP code into the connected consumer electronics devices.

The method comprises: the transmission unit of base station gets the transmission bandwidth information of Iub interface; the transmission bandwidth information of Iub interface is sent to DSP algorithm unit by using Iub interface transmission bandwidth allocation request message; according to the Iub interface transmission bandwidth information, DSP algorithm unit calculates each parameter allocated with HS-DSCH capacity so as to control the traffic of Iub interface under HSDPA mode.

The invention discloses a DWDM optical channel identification monitoring method and system based on a frequency domain perturbation optical label, relates to the technical field of optical communication. Only one group of receiving devices is required in one transmission direction so as to greatly reduce the cost and complexity of future optical network nodes. According to the invention, the DWDMoptical channel signal does not need to be demodulated at the optical network node, only spectral analysis needs to be carried out on the label signal, any effective information does not need to be loaded on the label signal, and meanwhile, the label signal does not need to be demodulated at the node, so that a complex DSP algorithm is avoided, and the system cost and complexity are reduced.