386 results about "Post filtering" patented technology

A computer system identifies malicious Uniform Resource Locator (URL) data items from a plurality of unscreened data items that have not been previously identified as associated with malicious URLs. The system can execute a number of pre-filters to identify a subset of URLs in the plurality of data items that are likely to be malicious. A scoring processor can score the subset of URLs based on a plurality of input vectors using a suitable machine learning model. Optionally, the system can execute one or more post-filters on the score data to identify data items of interest. Such data items can be fed back into the system to improve machine learning or can be used to provide a notification that a particular resource within a local network is infected with malicious software.

A user of a meta-search engine submits a query formulated with operators defining relationships between keywords. Information sources are selected for interrogation by the user or by the meta-search engine. If necessary, the query is translated for each selected source to adapt the operators of the query to a form accepted by that source. The query is submitted to each selected source and answers are retrieved from each source as a summary of each document found that satisfies the query. The answers are post-filtered from each source to determine if the answers satisfy the originally formulated query. Answers that satisfy the query are displayed as a list of selectable document summaries. The analysis includes computing a subsumption ratio of filtered answers to answers received that satisfy a translated query. The subsumption ratio is used to improve the accuracy of subsequent queries submitted by the user to the meta-search engine.

Face detection techniques are provided that use a multiple-stage face detection algorithm. An exemplary three-stage algorithm includes a first stage that applies linear-filtering to enhance detection performance by removing many non-face-like portions within an image, a second stage that uses a boosting chain that is adopted to combine boosting classifiers within a hierarchy “chain” structure, and a third stage that performs post-filtering using image pre-processing, SVM-filtering and color-filtering to refine the final face detection prediction. In certain further implementations, the face detection techniques include a two-level hierarchy in-plane pose estimator to provide a rapid multi-view face detector that further improves the accuracy and robustness of face detection.

An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.

An Ultraviolet-C (UVC) based portable water purification system employing a novel array of baffles increases the efficiency per unit energy of irradiating UVC light in the eradication of pathogens in the water. Closed loop feedback allows monitoring the application of UVC light power to ensure high levels of pathogen eradication. This system is capable of eradicating a wide range of waterborne bacteria, viruses, protozoa, helminthes, yeast, and mold found in natural freshwater sources worldwide. By adding pre- or post-filters, the system can remove harmful organic compounds, pesticides, inorganic compounds and heavy metals from the water. The system can also be used to eradicate pathogens in fluids other than water. As a feature of this invention, a communications systems that can reach geographically dispersed populations at low cost without the need to install costly wired communications infrastructure is combined with and powered by the water purification system. In one embodiment, a packet radio system is provided to create nodes in a wireless mesh communications system to provide voice, data, video and internet communications using an array of the water purifiers to create a wireless mesh network.

Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

A post-processing method for a speech decoder which outputs a decoded speech signal in the time domain provides high frequency resolution based on a frequency spectrum having non-harmonic and noise deficiencies. This is obtained by transforming the decoded time domain signal to a frequency domain signal by using a high frequency resolution transform (FFT). Then an analysis of the energy distribution of the frequency domain signal is made throughout its frequency area (4 kHz) to find the disturbing frequency components and to prioritize such frequency components which are situated in the higher part of the frequency spectrum. Next, the suppression degree for the disturbing frequency components is found based on prioritizing. Finally the steps of controlling a post-filtering of the transform in dependence of the finding, and inverse transforming the post-filtered transform in order to obtain a post-filtered decoded speech signal in the time domain are performed.

A post-filter includes a microphone array including at least two microphones to which a voice signal are input, a beam former which forms the voice signal input from the microphone array, a divider which divides a target sound containing noise input from the microphone array into at least two frequency bands at a predetermined frequency, a first filter which estimates the filter gain with the noise non-correlated between the microphones, a second filter which estimates a filter gain of one microphone of the microphone array or an average signal of the microphone array, an adder which adds the outputs from the first and second filters to each other, and a filter for reducing the noise based on the outputs from the adder and the beam former.

A “Sensor Array Post-Filter” provides an adaptive post-filter that accurately models and suppresses both diffuse and directional noise sources as well as interfering speech sources. The post-filter is applied to an output signal produced by a beamformer used to process signals produced by a sensor array. As a result, the Sensor Array Post-Filter operates to improve the signal-to-noise ratio (SNR) of beamformer output signals by providing adaptive post-filtering of the output signals. The post-filter is generated based on a generative statistical model for modeling signal and noise sources at distinct regions in a signal field that considers prior distributions trained to model an instantaneous direction of arrival for signals captured by sensors in the array.

Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

The invention provides an LED straight lamp. According to the LED straight lamp, the safety function that energizing is achieved only when two end covers of a lamp tube are put in a lamp holder of the lamp simultaneously is achieved, and the effect that installation personnel can avoid the danger of electric shock during installation is ensured. The LED straight lamp comprises the lamp tube, two lamp holders, a rectifying circuit, a filter circuit, an LED driving module and a detection module. The two lamp holders are each provided with a pin used for receiving an outside driving signal and are suitable for being connected with the two ends of the lamp tube correspondingly in a coupling mode. The rectifying circuit is used for rectifying the outside driving signal so as to generate a post-rectifying signal. The filter circuit is used for generating a post-filtering signal. The LED driving module is configured to generate a driving signal to make the LED straight lamp emit light. The detection module is used for determining whether the outside driving signal is stopped from flowing through the LED straight lamp and is provided with a first detection end and a second detection end. When an electric current flowing through the first detection end and the second detection end is higher than or equal to an electric current value, the detection module is switched on. When the electric current flowing through the first detection end and the second detection end is lower than the electric current value, the detection module is switched off.

Systems and methods of improved noise reduction using direction of arrival information include: receiving an audio signal from two or more acoustic sensors; applying a beamformer module to employ a first noise cancellation algorithm to the audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm; using spatial information derived from the audio signal received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal in proportion to the difference between the measured direction-of-arrival and the target direction-of-arrival; and outputting a single audio stream with reduced background noise.

An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

An audio encoder, an audio decoder or an audio processor includes a filter (12) for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal (16), the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller (18) is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor (22) having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

A method to improve packet loss concealment for generation of a synthetic speech signal in a algebraic code excited linear prediction decoder for a voice over packet network. One method improves features for coding gains in the decoder and for post-filtering of the signals. An alternative method uses a classification method for the signal based on the bitstream in the decoder.

The invention discloses a voice enhancement method employing a combination of nesting-subarray-based post filtering and spectrum-subtraction and is suitable for indoor environment, comprising the enhancement of multi-channel voice signal in vehicle environment; as the problems of unstable the broadband of the voice signals, the inconsistent frequency response of the microphone-array-based multi-channel voice-enhancement method to the voice signal and the correlation among all-channel noise in actual noise-field environment are considered, by utilizing the microphone array nested by the subarrays with different spacing, the voice signals are collected; and the voice signals formed by subarray beams are divided into a high-frequency section and a low-frequency section, different voice-enhancement algorism are adopted for carrying out the treatment; all the advantages are complementary with each other, thus improving the effect of voice enhancement.

A computer system identifies malicious Uniform Resource Locator (URL) data items from a plurality of unscreened data items that have not been previously identified as associated with malicious URLs. The system can execute a number of pre-filters to identify a subset of URLs in the plurality of data items that are likely to be malicious. A scoring processor can score the subset of URLs based on a plurality of input vectors using a suitable machine learning model. Optionally, the system can execute one or more post-filters on the score data to identify data items of interest. Such data items can be fed back into the system to improve machine learning or can be used to provide a notification that a particular resource within a local network is infected with malicious software.

In a mobile terminal device having a processing unit and a memory unit, an acoustic echo cancellation and a microphone beamforming are provided. Said device includes a plurality of a microphones, a pre-processing stage has an amplifier and a signal converter for an analog-to-digital conversion. In a main signal processing stage, a pre-filter suitable for polynomial beamforming, an acoustic echo cancellation stage and a post-filter for polynomial beamforming are provided. Furthermore, a post-processing stage has an amplifier and a signal converter for a digital-to-analog conversion, and plurality of speakers.

A microphone array system (16) for sound acquisition from multiple sound sources in a reception space surrounding a microphone array (18) that is interfaced with a beamformer module (28) is disclosed. The microphone array (18) includes microphone transducers (22) that are arranged relative to each other in N-fold rotationally symmetry, and the beamformer includes beamformer weights that are associated with one of a plurality of spatial reception sectors corresponding to the N-fold rotational symmetry of the microphone array (18). Microphone indexes of the microphone transducers (18) are arithmetically displaceable angularly about the vertical axis during a process cycle, so that a same set of beamformer weights is used selectively for calculating a beamformer output signal associated with any one of the spatial reception sectors. A sound source location module (30) is also disclosed that includes a modified steered power response sound source location method. A post filter module (32) for a microphone array system is also disclosed.

The invention provides a video coding post filtering method based on a convolutional neural network. The method comprises a convolutional neural network model training step and a filtering step, the training step comprises: setting a quantization parameter of video compression to 20 to 51, and performing coding compression on the original video to obtain a compressed video; performing frame extraction on all videos to obtain a plurality of frame pairs of compressed video frames and original video frames; dividing the extracted frame pairs into a plurality of groups according to the frame types and different quantization parameters; establishing a convolutional neural network framework and initializing the network parameters, and training the neural network by using the divided groups to obtain a plurality of convolutional neural network models corresponding to different quantization parameters and frame types. The filtering step comprises: executing the foregoing coding compression and frame extraction on the to-be-processed original video to obtain to-be-processed frame pairs, and selecting the corresponding convolutional neural network models according to the quantization parameters and the frame types of the to-be-processed frame pairs to perform filtering processing.

A digital sound system suitable for a digital speaker device for directly converting analog sound by a circuit using a ΔΣ modulator and a mismatch shaping filter circuit to output a plurality of digital signals and a plurality of speakers driven by the plurality of digital signals. A digital speaker driving device includes a ΔΣ modulator, a post filter, s driving circuits, and a power supply circuit to supply power to the ΔΣ modulator, the post filter and the s driving elements and the s driving circuits are adapted to s digital signal terminals.

A digital sound system suitable for a digital speaker device for directly converting analog sound by a circuit using a ΔΣ modulator and a mismatch shaping filter circuit to output a plurality of digital signals and a plurality of speakers driven by the plurality of digital signals. A digital speaker driving device includes a ΔΣ modulator, a post filter, s driving circuits, and a power supply circuit to supply power to the ΔΣ modulator, the post filter and the s driving elements and the s driving circuits are adapted to s digital signal terminals.

A motion picture depth information processing system for processing depth information of a plurality of frames in a motion picture segment, each of which has corresponding color information, is composed of a buffer, a motion estimation unit, a motion compensation unit, and an adaptive post-filtering unit. The system can carry out motion estimation by referring to the color information of the frames to obtain a motion vector for each block in a frame and depth motion compensation based on the aforementioned motion vectors to get an initial depth map for each frame in the segment, then do a post-processing via the adaptive post-filtering unit to get a refined depth map of every frame, and finally generate more definite depth information for the whole motion picture segment, thus reducing geometrical texturing burrs, which are commonly known in the depth processing prior art technology.

The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, preferably as the area below the square of the magnitude of the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

The invention discloses an echo elimination method based on deep learning, an echo elimination device based on deep learning, electronic equipment and storage medium, and belongs to the technical field of a computer. The method comprises the following steps of extracting acoustic features from received microphone signals, wherein the microphone signals comprise near-end signals and remote-end signals; performing iterative operation on the acoustic features in a pretrained recursive neural network model with long short-term memory; calculating a ratio membrane of the acoustic features; using the ratio membrane for shielding the acoustic features; and synthesizing the shielded acoustic features with the phase of the microphone signals to obtain near end signals subjected to echo elimination.The echo elimination method based on deep learning and the echo elimination device based on deep learning can realize the echo elimination under the conditions of the background noise, double talking, nonlinear distortion and the like; the echo elimination effect is greatly improved; the application scenes are greatly increased; the use of a post-filter is not needed; the electronic equipment iseffectively simplified; and the electronic equipment cost is reduced.

An Ultraviolet-C (UVC) based portable water purification system employing a novel array of baffles increases the efficiency per unit energy of irradiating UVC light in the eradication of pathogens in the water. Closed loop feedback allows monitoring the application of UVC light power to ensure high levels of pathogen eradication. This system is capable of eradicating a wide range of waterborne bacteria, viruses, protozoa, helminthes, yeast, and mold found in natural freshwater sources worldwide. By adding pre- or post-filters, the system can remove harmful organic compounds, pesticides, inorganic compounds and heavy metals from the water. The system can also be used to eradicate pathogens in fluids other than water. As a feature of this invention, a communications systems that can reach geographically dispersed populations at low cost without the need to install costly wired communications infrastructure is combined with and powered by the water purification system. In one embodiment, a packet radio system is provided to create nodes in a wireless mesh communications system to provide voice, data, video and internet communications using an array of the water purifiers to create a wireless mesh network.

Systems and methods of improved noise reduction include the steps of: receiving an audio signal from two or more acoustic sensors; applying a beamformer to employ a first noise cancellation algorithm; applying a noise reduction post-filter module to the audio signal including: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm, wherein the current noise spectrum is estimated using the audio signal received by the second acoustic sensor; determining a punished noise spectrum using the time-average level difference between the audio signal received by the first acoustic sensor and the current noise spectrum; determining a final noise estimate by subtracting the punished noise spectrum from the current noise spectrum; and applying a second noise reduction algorithm to the audio signal received by the first acoustic sensor using the final noise estimate; and outputting an audio stream with reduced background noise.