321 results about "Noise intensity" patented technology

This invention is related to a remote and digital data transmission system and satellite location from mobile or fixed terminals with urban surveillance cameras for facial recognition, detector of gunshots, data collection of public security personnel and missing or kidnapped individuals, city alarms and vehicles control collecting traffic violations and detection of stolen vehicles, all modules communicated to a command center comprising the combination of a close-up camera placed on a post or structure, camera by means of which a photograph is taken, said photograph is sent to the command center (2) by means of a data link providing the photograph (3), in real time, said photograph is sent to a global module (4) which transmits it to a security filter (5) for its analysis and once the information is confirmed it sends it back to the module (4), and from here it is sent to the individuals identification module (6), and it edits the photograph or card with all data (7) with an arrangement of a series of microphones (8) coupled to a structure and an electronic switchboard (9), said arrangements enable the recognition of sound signatures such as gunshots in different noise intensities, and said command center acts as an information CPU to register the payment of fines by means of automatic traffic violations and different public services addressed to the corresponding individuals by means of a control system of sole identity with capacity in addition for application of payments.

The invention discloses a self-adapting random resonance weak signal detection method, which achieves the self-adapting detection of weak periodic signal through combining a matching random resonance and an imposed signal induced random resonance. According to the theory of random resonance, firstly, numerical simulation is processed to receive correspondence between the noise intensity and the frequency of the best matching random resonance of a nonlinear bistable system in a range of extremely low frequency and under the condition of small-signal; and a high-frequency weak signal encountered in a project can be processed by using the change of normalized scale. For the detection of a weak signal with extremely low message-to-noise ratio and large noise intensity, the heights of voltage barriers of the bistable system can be changed by using the induced random resonance, thereby helping a weak eigenfrequency signal to fulfill the transition between voltage barriers, so that the bistable system reaches the random resonance state, thereby achieving the self-adapting detection of the weak periodic signal.

The invention discloses an automatic volume adjustment method. The method comprises the following steps of: recording external sound by an input unit; converting the external sound into a digital signal; identifying whether the digital signal is ambient noise or user language; if the digital signal is the ambient noise, judging the intensity of the ambient noise; acquiring volume adjustment corresponding to the intensity of the ambient noise from a comparison table of noise intensity and volume adjustment, wherein the comparison table defines that the volume of received sound of the other party of a call is turned up along with the increase of the intensity of the ambient noise; and adjusting the volume of the received sound of the other party of the call according to the acquired volume and outputting the sound. The invention also provides an automatic volume adjustment device. By using the automatic volume adjustment device and the automatic volume adjustment method of the invention, a user can receive the audio frequency of the other party of the call without manually adjusting the volume.

The invention discloses a method for adjusting volume output of a mobile terminal according to environment noises and the mobile terminal based on the method. The method includes real-timely detecting conditions of the environmental noises around the mobile terminal for data collection, analyzing collected noise intensity, comparing the noise intensity with parameter grades of the environmental noises, outputting parameter values for adjusting the volume of output audio, and controlling a telephone receiver of the mobile terminal to output corresponding volume. The mobile terminal comprises a preset module, an environment noise input module, an audio output adjusting module, and an environment noise analyzing and comparing module. By means of the method and the mobile terminal, by adjusting output volume of a device in different background noises, users all can clearly hear audio output of the device under conditions of different volume background noises, and the problem that people around the users can hear sounds in microphones of the users in a silent environment due to too loud sounds in the microphones can be avoided.

The invention discloses an adaptive noise intensity video denoising method which is based on motion detection and is embedded in an encoder. The method comprises the following steps: (1) taking a sum of regularization frame differences in a neighborhood as an observed value, dividing input pixels into a static pixel and a dynamic pixel and using filters in different supporting domains for the two kinds of the pixels, wherein a filtering coefficient is adaptively determined according to noise intensity and an image local characteristic; (2) taking a single DCT coefficient or the sum of the several DCT coefficients as the characteristic, using AdaBoost as a tool to construct a cascade-form classifier and using the classifier to select a static block; (3) establishing a function model of connection between DCT coefficient distribution parameters of the video noise intensity and the static block and using the model to estimate the noise signal standard difference. By using noise intensity estimation embedded in the video encoder and a noise reduction technology provided in the invention, few computation costs can be used to acquire the parameters and the information needed by noise filtering. A time efficiency is good. Because a reliable clue is used to determine whether the pixels accord with a static hypothesis, the filter of the invention can effectively filter the noise and simultaneously maintain marginal sharpness of the static image. And motion blur caused by filtering in a motion area can be avoided.

The invention discloses an environment noise wireless real-time monitoring system based on a ZigBee technology, mainly comprising a noise monitoring control device and more than one terminal noise monitoring device, wherein the terminal noise monitoring device and the noise monitoring control device are connected by a wireless communication mode. The environment noise wireless real-time monitoring system based on the ZigBee technology can be used for monitoring municipal environment noise, and has broad application prospect; noise intensity analysis is carried out in a dispersed mode at each terminal noise monitoring device, the treatment pressure and system communication expense of the noise monitoring control device are reduced, and the system data treatment delay and transmission delayare reduced to realize real-time noise monitoring; and a ZigBee noise monitoring network has the characteristic of self organization, and the terminal noise monitoring devices are easy to join the network, thereby greatly facilitating the extension of system capacity.

The invention discloses an audio spread spectrum watermark processing method for protecting the audio data, comprising two processes: a watermark embedding process and a watermark extraction process; in the embedding process, a wavelet de-noising technique is used for separating the noise of the signal and noise intensity control factor is used for reducing the noise signal intensity, and the effect of the noise of the signal on the detector performance is reduced; in order to improve the performance of the detector, the wavelet de-noising technique is also used in the water extraction process; in order to overcome the problem of excessively large correlation sliding detection calculation quantity in the corresponding detection, a synchronous positioning mechanism is designed. The algorithm is applicable to music and speech, which can effectively resist the attack of normal signal processing, more particularly, the attack of DA / AD conversion, thus being applied to the fields such as audio copyright protection, temper-proofing, certification, etc.

The invention discloses a de-noising method of filtering images in size adaptive block matching transform domains. Less false signals are introduced as two-dimension transformation of each image block in the block matching 3D (BM3D) in a basic estimation stage is abandoned by the method; image details can be well preserved as the block number in blocking matching groups of the method is less than the block number in the BM3D method. The image de-noising performance of the method is further improved as the method adaptively selects the block size based on form components during block matching. The current general objective evaluation of image de-nosing includes peak signal noise ratio (PSNR) and mean structural similarity (MSSIM), and according to the method, the de-noising calculation results of a plurality of standard images provided on BM3D networks are higher than the results of the BM3D method on the basis of the two objective evaluations and under all noise intensities.

The invention provides a ship structure vibration and noise forecasting system based on S-P-R, comprising a database module, a modeling and calculating module, and a result processing module. The database module stores the vibration intensity data of various kinds of vibration sources in each cabin of a ship, and the noise intensity data of various kinds of noise sources; the modeling and calculating module comprises a ship modeling submodule, a key excitation source recognition submodule, a transmission path determination submodule, and a receiving station energy calculating module, and is used for defining transmission paths, calculating the transmission loss of the vibration / noise energy generated by each excitation source by the transmission to a receiving station along a transmission path, and then calculating the vibration / noise total energy of the receiving station. The system of the invention combines an analysis method, a numerical method and an experimental method to analyze ship structure vibration source and transmission path characteristics, calculates cabin air noise levels according to the analysis procedure of an S-P-R method, and overcomes the problem that corresponding remedial measures can not be taken according to vibration and noise measurement results until a ship is completed.

The invention discloses a voice recognition method, a mobile terminal and a computer readable storage medium. The method comprises the steps of detecting whether the noise intensity in the current environment meets a preset condition in a voice communication process, acquiring a lip feature image if the noise intensity meets a preset condition, recognizing lip language information according to thelip feature image, converting the lip language information into voice information and / or text information and sending the voice information and / or text information to a voice communication receivingend. Accurate voice information and / or text information is received by the voice communication receiving end in a noisy environment, so that the communication quality is improved.

The invention relates to the field of video processing, and discloses a color image denoising method and a system thereof. In the method and the system thereof, cross-channel invariable information is extracted from a luminance channel image for the denoising treatment on a chrominance channel image, so that a false color edge or texture can be eliminated, the fuzzyness of the edge part can be reduced, and better integral denoising effect can be obtained. The cross-channel invariable information can be edge information, texture information, noise intensity and the like. The luminance channel may be firstly processed with local data conversion; the cross-channel invariable information is extracted from a conversion domain; and characteristic information, such as edge texture and the like can be extracted from local image data more effectively based on different characteristics of a signal represented in the conversion domain, and consequently, the chrominance channel image can be denoised better based on the information.

The invention discloses a target tracking method of a passive multi-sensor based on layered particle filtering. A sub-state including an azimuth angle, a change ratio of the azimuth angle, a change ratio of a logarithm radial distance and auxiliary parameters and a sub-state Psi including the logarithm radial distance are constructed to realize a structure of hierarchical filtering through rewriting a system equation of a logarithm polar coordinate and adding the auxiliary parameters indicative of process noise intensity and radial distance ratio, wherein a first layer updates the auxiliary parameters in a second layer by means of sequential importance sampling method according to observation information of various sensors; Psi is iterated and updated; and the auxiliary parameters are combined to obtain estimation of the target state; finally a fusion output result of the target state is obtained according to an optimal information fusion method. Values of the auxiliary parameters can be estimated in real time by using a method of the layered particle filtering, and errors introduced by a maximum value of the noise intensity in the use process of the filtering is avoided, such that the problem of the target tracking can be effectively solved under the conditions of unknown process noise intensity and unmeasured distance.

Embodiments of the invention provide a volume control method and a terminal thereof and relate to the computer technology field. By using the method and the terminal, the volume can be adaptively adjusted according to changes of an external environment. The method comprises the following steps of acquiring noise correlation information of the external environment where a user terminal is currently located; when a noise intensity of the external environment where the user terminal is currently located is greater than a noise threshold, inquiring a volume control template which is matched with the noise correlation information of the external environment where the user terminal is currently located; when the matched volume control template is inquired, determining the volume after adjustment needed by the user terminal according to the matched volume control template; using the volume after adjustment needed by the user terminal to adjust the volume of the user terminal. The method and the terminal in the embodiments of the invention are applied in a volume control process of the user terminal.

The invention discloses a method for mobile phone source identification an additive noise environment based on the constant Q transform domain, which is characterized in that a spectrum distribution characteristic vector of the constant Q transform domain is obtained by using constant Q transform, wherein the constant Q transform has higher frequency resolution at a low frequency and has higher time resolution at a high frequency, thereby being more suitable for mobile phone source identification; a multi-scene training mode is adopted in the training stage, there are not only clean voice samples without scene noise but also noisy voice samples with different scene noise types and noise intensities in a training set, an M-classification model obtained by the training is universal and can carries out effective mobile phone source identification on voice samples with known noise scenes and unknown noise scenes; and a CNN model of deep learning is used to establish the M-classification model, the CNN model not only improves the accuracy of source identification for the clean voice samples without scene noise but also greatly improves the effect of mobile phone source identification for the noisy voice samples, and the noise robustness is strong.

The present invention discloses an enhancement processing method and apparatus for a traffic video image. According to the method and the apparatus, the noise intensity is estimated and an image noise reduction process is guided according to continuous video image inter-frame information, a constraint weight of a neighborhood pixel region to a center pixel unit is introduced according to extracted local feature effective data, and a grey-scale mapping relationship function of the pixel unit is adaptively constructed, so that image noise suppression is more accurate, image local detail enhancement is more obvious, and further the effectiveness of a video image enhancement method in a complex scenario, especially a low-illumination environment is improved.

The invention discloses a strain field subarea dynamic selection method in a digital image correlation method, and provides evaluation parameters based on displacement field local gradient intensity for describing intensity of local non-uniform deformation. By balancing the noise intensity of the displacement field and the local non-uniform deformation intensity, a large strain calculation sub-region is selected at a place with relatively low non-uniform deformation gradient intensity, so that the noise of the displacement field is effectively suppressed; and a small strain calculation sub-region is selected in a place with relatively high non-uniform deformation gradient strength, so that the local deformation characteristic is reserved. And dynamic selection of the size of the strain field calculation subarea is realized, so that local deformation characteristics are reserved to the greatest extent while noise is effectively suppressed. The dynamic selection process of the size of the subarea based on the characteristics of the displacement field is different from the traditional dynamic selection process of calculating the size of the subarea based on the speckle gray level gradient information, and the measurement precision of the strain field of the digital image correlation method in the non-uniform deformation can be effectively improved.

The invention discloses a GPS weak signal tracking system based on I / Q branch correlation integral observation filtering. A receiver is used for receiving a satellite signal which is converted into an intermediate frequency signal through down conversion and transmitted to a frequency mixer. The frequency mixer is also used for receiving local sine and cosine reproduction carrier signals outputted by a local carrier digital-controlled oscillator, and outputting a I-branch output signal and a II-branch output signal to a correlation arithmetic unit. The correlation arithmetic unit also receives a local advanced C / A code, an instant reproduction C / A code and a lagged reproduction C / A code, wherein the local advanced C / A code, the instant reproduction C / A code and the lagged reproduction C / A code are generated by a code generator. Results are outputted to an integral eraser, and six-way relevant integral values are generated and outputted to a Kalman filter. The Kalman filter obtains an estimated carrier phase difference, an estimated carrier frequency difference and an estimated code phase difference, and the estimated carrier phase difference, the estimated carrier frequency difference and the estimated code phase difference are transmitted to the local carrier digital-controlled oscillator and the code generator. The system effectively reduces the noise intensity of the satellite signal, can trace the satellite signal better in a weak-signal environment, and improves the tracing precision.

A volume control device comprises an audio frequency amplifier, a sound collector, a weighter, a comparator, a storer and a central processor, wherein the audio frequency amplifier is used for outputting an amplified audio signal; the sound collector is used for detecting various sound sources in environment except for the audio signal and the intensities of the various sound sources; the weighter is used for acquiring a total noise intensity after superposition of the various sound sources; the comparator compares the difference between the obtained total noise intensity and the audio signal intensity; the storer is used for storing a gain table; the central processor is used for acquiring the gain adjustment amount of the audio frequency amplifier according to the difference and the gain table, and correspondingly increasing or decreasing the gain of the audio frequency amplifier according to the gain adjustment amount so as to correspondingly adjust the volume of the audio signal. The invention further relates to a volume control method.

The invention provides a monolingual sound source DOA estimation method based on an AVS and sparse representation. According to the method, the acoustic vector sensor is adopted to collect audio signals, high signal-to-noise ratio time frequency points are extracted by utilizing the trigonometric function relationship between the time frequency sparse characteristic of the voice signals and the receiving component of the AVS and adopting a sine track algorithm, and the ratio of data of a pressure gradient sensor to data of an all-around pressure sensor is calculated. On the basis, an over-complete dictionary sparse representation model for the ratio of data of the sensors is acquired through the spatial sparse characteristic of a sound source, and the monolingual sound source direction-of-arrival estimation problem is converted into a sparse vector solving problem. The sparse vector is solved with the l1-SVD method, the spatial power spectrum is calculated, and the DOA of the monolingual sound source can be obtained through calculation. The DOA of the monolingual sound source can be accurately estimated under conditions of different noise intensities and room reverberations with the method. Besides, a microphone array adopted for the method is small in size and only has the area of 1 cm<3>, and thus the method is very suitable for the voice technology for portable devices.

The invention provides a method for controlling volume output of a terminal loudspeaker and a mobile terminal, and relates to the field of electronic technology. The method aims at addressing the problem of poor user experience due to the fact that a user often has to manually adjust the volume of a loudspeaker when the volume is not appropriate each time the terminal turns on the loudspeaker. The method is applied to terminals. The method comprises the following steps: when the terminal turns on the loudspeaker and operates volume output, detecting the current noise intensity of ambient environment whereby the terminal is; based on the prestored appropriate volume data corresponding to different noise intensity, acquiring an appropriate volume corresponding to the current noise intensity; the appropriate volume data including a contrasting relationship table between a plurality of groups of historic loudspeaker volume data and / or different noise intensity corresponding to different noise intensity and a theoretical best loudspeaker volume; based on the appropriate volume, determining the volume of the terminal loudspeaker and outputting the volume. According to the invention, the method realizes that once the terminal is turned on, the loudspeaker voice can achieve the ideal sound effect which is adapt to noisy level of the ambient environment, which increases user experience and increases practicality and convenience.

Process and measuring equipment for improving the signal resolution in gas absorption spectroscopy, wherein the measuring equipment includes a laser light source, a light detector and a measuring chamber arranged in between, and furthermore a light source control unit and a light detector evaluation unit. To improve the signal resolution, the noise intensity of the measuring equipment is reduced by averaging over time the interfering signal portions caused by back-reflections, etalons respectively self-mixing effects. This is accomplished by a light modulator arranged downstream the laser light source that continuously periodically influences the optical path length of the light beam. Thereto the light modulator includes an optical element with an adjustable refractory index that continuously cyclically alters the phase of the laser light of the light beam.

A voice input device includes a first microphone (710-1) that includes a first diaphragm, a second microphone (710-2) that includes a second diaphragm, and a differential signal generation section (720) that generates a differential signal that indicates a difference between a first voltage signal and a second voltage signal, the first diaphragm and the second diaphragm being disposed so that a noise intensity ratio is smaller than an input voice intensity ratio (input voice component intensity ratio), and the differential signal generation section (720) including a gain section (760) that amplifies the first voltage signal by a predetermined gain, and a differential signal output section (740) that generates and outputs a differential signal that indicates a difference between the first voltage signal amplified by the gain section and the second voltage signal.

A noise generator includes a noise intensity storage means for storing a noise intensity threshold of a noise, a noise intensity setting means for setting an optimum noise intensity based on the noise intensity threshold, a noise generation means for generating the noise based on the optimum noise intensity, and a noise output means for outputting the noise generated by the noise generation means. The noise generator determines the noise intensity threshold based on perception of the noise by a driver of an automotive vehicle.

The invention discloses a method and device for demodulating pilot-tone modulation signals. The method comprises the following steps of: calculating an intensity value at a frequency position of each received pilot-tone modulation signal in a subsection manner according to a set time period; judging a signal intensity range and a noise intensity range at the frequency position of the received pilot-tone modulation signal at each time period according to the received intensity value during each time period; matching each intensity value with the intensity range within the corresponding time period to obtain the intensity range of each intensity value so that a bit value corresponding to each intensity value is demodulated. The device comprises an intensity value calculation module, an intensity range obtaining module and a bit information obtaining module. According to the method and device for demodulating the pilot-tone modulation signals, the pilot-tone modulation signals are demodulated regardless of the distribution state of noise by using the intensity range of dynamic regulation noise and the intensity range of the pilot-tone modulation signals; furthermore, the pilot-tone modulation signals are demodulated by using a corresponding relation between the same frequency of the pilot-tone modulation signals and a wavelength.

The invention discloses a bottom die and die set link mechanism of a bottle blowing machine, which ensures that the opening and the closing of a die set and the ascending and the descending of a bottom die are integrated, namely the bottom die descends or ascends when the die set is opened or closed. The link mechanism has the effects of improving the matching precision between the bottom die and the die set in the working process, also simplifying equipment, saving the used space, reducing the cost, effectively reducing the noise intensity, prolonging the service life of a roller and avoiding huge impact caused by the fact that the roller is in instant contact with a corner of a lifting guide rail of the bottom die when the equipment rotates at a high speed.

The invention discloses a link mechanism for opening and closing a die set and lifting a bottom die of a bottle blowing machine. The link mechanism comprises an opening-closing die connecting rod for driving the die set of the bottle blowing machine to open and close, wherein the opening-closing die connecting rod can be shaken under drive of a driving device; one end of a joint bearing connecting rod is connected with the opening-closing die connecting rod, while the other end of the joint bearing connecting rod is connected with a sliding block, wherein the sliding block can be mounted on a die set bracket in a horizontal and sliding manner; a driving seat is fixed on the sliding block; a bottom die lifting guide board with a height difference curve sliding way is fixed on the bottom die bracket for driving a bottom die to lift; and the driving seat is connected with a driving shaft, wherein the end part of the driving shaft can be extended into the curve sliding way and can slide in the curve sliding way. The link mechanism has the following beneficial effects: 1, size of the bottle blowing machine can be effectively reduced and a space utilization rate of a manufacturing shop can be improved; 2, noise intensity can be greatly reduced and service lives of corresponding parts can be prolonged; and 3, matching precision can be guaranteed easily and difficulty in manufacture, technology and installation of the corresponding parts can be reduced; and production cost can be reduced.

The invention provides a sound-intensity-method-based noise calculation method and system for a transformer multipoint sound source model. The method comprises the steps that noise intensity levels of a plurality of measure points on the side faces of a transformer are measured to obtain the total A-weighting sound intensity level of each measure point; noisy data is grouped according to the number of the measure points on each side face and the number of point sound sources needing to be equivalent; a plurality of groups of noisy data are subjected to superposition according to a sound intensity method to obtain total A-weighting sound power levels of the multiple equivalent point sound sources; the multiple equivalent point sound sources on each side face form a face sound source of the side face, wherein the positions of the multiple equivalent point sound sources are allocated according to positions of the measure points; a multipoint sound source model for multiple equivalent points of each side face of each phase of the transformer is established; the position of a predicted point is determined, and the equivalent point sound sources for the noise at the position of the predicted point are determined according to the multipoint sound source model; coordinate systems are established for predicted points in different directions, and the noise at the predicted points is calculated according to the point sound source outdoor transmission theory.

The invention provides a voice input device, its manufacturing method and information processing system. The voice input device includes a first microphone (710-1) with a first vibrating membrane, a second microphone (710-2) with a second vibrating membrane, and a difference signal generating unit (720) that generates a difference signal between first and second voltage signals. The first and second vibrating membranes are disposed to make a noise intensity ration smaller than an input voice intensity ratio indicative of a ratio of the intensity of input voice components. The difference signal generating unit (720) includes a delay unit (730), and a difference signal output unit (740) that generates a difference signal between delayed signals to which the delay unit gives delays and that outputs the difference signal.