334 results about "Word error rate" patented technology

Memory cells are programmed and read, at least M=3 data bits per cell, according to a valid nonserial physical bit ordering with reference to a logical bit ordering. The logical bit ordering is chosen to give a more even distribution of error probabilities of the bits, relative to the probability distributions of the data error and the cell state transition error, than would be provided by the physical bit ordering alone. Preferably, both bit orderings have 2M−1 transitions. Preferably, the logical bit ordering is evenly distributed. The translation between the bit orderings is done by software or hardware.

Non-volatile solid-state memory-based storage devices and methods of operating the storage devices to have low initial error rates. The storage devices and methods use bit error rate comparison of duplicate writes to one or more non-volatile memory devices. The data set with a lower bit error rate as determined during verification is maintained, whereas data sets with higher bit error rates are discarded. A threshold of bit error rates can be used to trigger the duplication of data for bit error comparison.

Low-Density Parity-Check (LDPC) codes offer error correction at rates approaching the link channel capacity and reliable and efficient information transfer over bandwidth or return-channel constrained links with data-corrupting noise present. They also offer performance approaching channel capacity exponentially fast in terms of the code length, linear processing complexity, and parallelism that scales with code length. They also offer challenges relating to decoding complexity and error floors limiting achievable bit-error rates. Accordingly encoders with reduced complexity, reduced power consumption and improved performance are disclosed with various improvements including simplifying communications linking multiple processing nodes by passing messages where pulse widths are modulated with the corresponding message magnitude, delaying a check operation in dependence upon variable node states, running the decoder multiple times with different random number generator seeds for a constant channel value set, and employing a second decoder with a randomizing component when the attempt with the first decoder fails.

A clock recovery circuit has a boundary detection / discrimination circuit to detect and discriminate a boundary in an input signal in accordance with a first signal. The clock recovery circuit performs clock recovery by controlling the timing of the first signal in accordance with the detected boundary, wherein boundary detection timing in the boundary detection / discrimination circuit is varied by controlling the first signal.

The invention relates to the coding of speech at a variable bit rate, whereby the bit rates can vary from frame to frame, and more specifically to the methods and filters used for improving the quality of the decoded speech. In the solution according to the invention the weighting factors of the postfilter are not adapted on the basis of the momentary bit rate or the bit rate used in the coding of each frame, but the weighting factors are adapted according to the average bit rate calculated on the basis of a predetermined length of time. In addition to this, the weighting factors of the postfilter are also adjusted on the basis of whether the frame in question contains a voiced speech signal, an unvoiced speech signal or background noise. At frames containing an unvoiced speech signal or background noise, postfiltering is weakened so as to avoid the distortion of the signal tone because the postfiltering is adapted to a voiced signal. The weighting factors of the postfilter can also be adapted on the basis of the error rate or other parameter describing the quality of the signal or the data transfer channel. For example, postfiltering can conveniently be adjusted so that when the channel error rate and the amount of coding error increase, postfiltering is increased, whereby the effect of data transfer errors on the decoded speech signal is reduced and the tolerance of the system with regard to data transfer errors increases.

The invention provides a neural network equalization method used for an indoor visible light communication system, and belongs to the visible light wireless communication technology field. The method includes the steps: utilizing a ceiling bounce model to calculate a VLC channel impulse response, carrying out photoelectric conversion for a visible light power signal received by a receiving end, and sending a sequence to a neutral network channel equalizer after amplification sampling; utilizing a heredity algorithm to optimize initialization weights and thresholds among neurons, establishing a neural network for training, and minimizing an error function; and judging an output, restoring the sent sequence, and finally achieving an equalization purpose. According to the scheme, interference among codes is obvious minimized, an error rate is reduced, the communication quality is further improved, a transmission rate that the system can reach is increased, the training time is shortened, and the system complexity is reduced.

A disk drive utilizes a unique write condition for each of the transducers within the drive. Each write condition is determined based upon the specific properties of the corresponding transducer. The write condition information is preferably stored within a memory within the disk drive. When a write operation is performed, the appropriate write condition for the corresponding transducer is used to determine when to write data to the disk. A write condition will typically include one or more individual write criteria. For example, a write condition can specify a write threshold value to be used during a write operation. A write fault threshold value is selected for each transducer by determining a position error distribution corresponding to that transducer wherein the position error distribution is essentially a function of random noise associated with the disk drive, and by using that distribution to determine allowable off-track threshold value (WFL), will protect the drive from encroachments.

Two statistics are disclosed for determining the quality of language models. These statistics are called acoustic perplexity and the synthetic acoustic word error rate (SAWER), and they depend upon methods for computing the acoustic confusability of words. It is possible to substitute models of acoustic data in place of real acoustic data in order to determine acoustic confusability. An evaluation model is created, a synthesizer model is created, and a matrix is determined from the evaluation and synthesizer models. Each of the evaluation and synthesizer models is a hidden Markov model. Once the matrix is determined, a confusability calculation may be performed. Different methods are used to determine synthetic likelihoods. The confusability may be normalized and smoothed and methods are disclosed that increase the speed of performing the matrix inversion and the confusability calculation. A method for caching and reusing computations for similar words is disclosed. Acoustic perplexity and SAWER are determined and applied.

A network receiver is dynamically configurable to process access bursts received from mobile stations as both normal-type and abbreviated-type access bursts. With this capability, radio signal measurement data may be collected in the network cell or sector in which the receiver is positioned for signals originating from mobile stations operating in surrounding cells, whether or not those surrounding cells use the same burst structure. In an exemplary implementation, the receiver determines whether access burst data exhibits a lower error rate when processed as a normal burst, or when processed as an abbreviated burst. The data derived from the processing approach that yields the lower error rate is then compared with an error threshold defining an unacceptable error level for the burst data. If the data is below the threshold, it is retained for post-processing analysis, where it is used to develop accurate inter-cell measurement data for the network.

A method and apparatus for channel coding useful for recording channel and other communications applications. The proposed channel coding method is actualized via structured set partition (SSP) in conjunction with multilevel coding (MLC) and offers performance gains over conventional coding schemes with comparable complexity at the bit-error rate (BER) level as well as the sector failure rate (SFR) level.

Alternate methods, devices and systems are presented for providing service providers and / or end users of mobile stations to monitor and / or report regions with high error rates and / or dead zones. A protocol is provided such as to yield the error occurrence, error rate and / or the address of the end user. In some embodiments it also contains location information and time of day information. This information may be collected from the G.P.S., and / or all or a portion of users when there is a significant transmission difficulty and / or an abundance of errors. The collected data forms a database which is analyzed to generate coverage maps. This may include the dynamic aspects of the mobile network. In another aspect of this invention, each mobile station periodically compares its current location with the data base. The results of this comparison enables the mobile unit to anticipate connection problems. In still another aspect of the invention, the comparison is performed by the service provider which notifies the end user.

Programmable logic devices are provided having configuration memory cells that exhibit decreased soft error rates. A stabilizing capacitor may be connected between each of the memory cell's input and output terminals. The capacitor may be a metal-insulator-metal capacitor formed using a vertical structure, a horizontal structure, or a hybrid vertical-horizontal structure. The memory cell may have inverter transistors of increased strength to help stabilize the memory cell.

Systems and methods are disclosed herein for improving the sensitivity of satellite data decode in a satellite navigation receiver. The low signal ephemeris data decoding system of the present disclosure achieves a 5 db improvement in decoding sensitivity over conventional system by operating down to a CN0 of 21 dB-Hz. The improved sensitivity is achieved through a combination of reducing the number of data bits to be decoded, overcoming the inherent differential decoding problem of an all data bit polarity inversion, improving the probability of seeing single bit decoding error in an ephemeris word, running the parity correction algorithm, and reducing the undetected word error rate. The improved sensitivity makes it possible to predict the orbit of the satellite and to determine the receiver's location with higher accuracy even when operating in challenging signal conditions.

Methods and apparatus are provided for sequential authentication of a user that employ one or mole error rates characterizing each security challenge. According to one aspect of the invention, a user is challenged with at least one knowledge challenge to obtain an intermediate authentication result; and the user challenges continue until a cumulative authentication result satisfies one or more criteria. The intermediate authentication result is based, for example, on one or more of false accept and false reject error probabilities for each knowledge challenge. A false accept error probability describes a probability of a different user answering the knowledge challenge correctly. A false reject error probability describes a probability of a genuine user not answering the knowledge challenge correctly. The false accept and false reject error probabilities can be adapted based on field data or known information about a given challenge.

Data communication between two devices (11, 12) over the public powerline network (13) uses a multi carrier technology wherein a communication channel is divided into a plurality of subchannels by frequency division multiplexing. A partial power as part of the total transmission power and a partial rate as a fraction of the total data rate are assigned to each subchannel as follows:Initially, the same partial power is assigned to each subchannel and the S / N ratio of each subchannel is obtained. Starting with the subchannel of the lowest S / N ratio and proceeding to that of the highest S / N ratio, the following steps are performed for each subchannel:Firstly, the partial rate is assigned in accordance with the respective S / N ratio such as to result in a predetermined transmission error rate. The assigned partial rate is then quantised to an integral value. If the quantising corresponds to a rounding down, the initially assigned partial power is reduced such that the transmission error rate remains the same as before quantisation. Further, the partial power of other subchannels with higher S / N ratio is increased such that the sum of all partial powers remains constant. Eventually, the increased S / N ratio resulting from increasing the partial power for these subchannels is calculated so that, when the present steps are conducted for them, an optimum partial rate can be assigned to them.This method maximises the total data rate of the entire communication channel while the error rate of each subchannel and the sum of all partial powers remain constant.

A method and system are provided for accelerated detection of soft error rates (SER) in integrated circuits (IC's) due to transient particle emission. An integrated circuit is packaged for accelerated transient particle emission by doping the underfill thereof with a transient-particle-emitting material having a predetermined emission rate. The emission rate is substantially constant over a predetermined period of time for testing. Accelerated transient-particle-emission testing is performed on the integrated circuit. Single-event upsets due to soft errors are detected, and a quantitative measurement of SER is determined.

A structure and a method for reduction of soft error rates in integrated circuits. The structure including: a semiconductor substrate; and a stack of one or more wiring levels stacked from a lowermost wiring level to an uppermost wiring level, the lowermost wiring level nearer the semiconductor substrate than the uppermost wiring level; and an alpha particle blocking layer on a top surface of the uppermost wiring level of the one or more wiring levels, the blocking layer comprising metal wires and a dielectric material, the blocking layer having a combination of a thickness of the blocking layer and a volume percent of metal wires in the blocking layer sufficient to stop a predetermined percentage of alpha particles of a selected energy or less striking the blocking layer from penetrating into the stack of one or more wiring levels or the substrate.

The invention discloses a method of modulating a variable pulse amplitude position and improving an error rate for a visible light communication system, belonging to the field of visible light wireless communication technologies. According to the method, modulation of the variable pulse amplitude position is utilized in a modulation module of a signal transmitting end, a modulation method is multi-system modulation capable of combining pulse amplitude position modulation and pulse width modulation, transmission of communication data is guaranteed while a system dimming function is provided, improvement of the communication quality mainly includes that the transmission rate can be improved under the condition of the same symbol rates, the advantages of high multi-system modulation bandwidth efficiency and high pulse position modulation power efficiency are combined, the spectral efficiency can be effectively improved, and compared with the other multi-system modulation manner, the complexity is lower. In order to further improve the error rate of the visible light communication system, the method utilizes a channel balancer based on a feed-forward back propagation multilayer perceptron on a receiving end, so that the error rate is effectively improved, and the communication quality is further improved.

In an optical communication system, a train of optical pulses is transmitted such that each pulse carries a set of multiplexed channels. In a receiver, the error rate for each channel is monitored and a subset of channels having the most favorable rate is selected to carry a first data stream. The remaining channels carry a second data stream. Error rate performance is determined on the basis of error detecting codes carried by the channels in the first and second data streams.

A computing device configured for wireless communication may effectively control adaptation to channel conditions. The device may be configured to identify and classify conditions impacting performance of a channel so that appropriate adaptations may be made. Interference may be detected by correlating received signal strength and packet errors. High received signal strength correlated to a high packet error rate may signify presence of a source of interference. Once a source of interference is detected, other criteria may be used to determine the nature of the interference so that an adaptation that is minimally disruptive of applications can be selected. Additionally, channel degradation may be predicted by monitoring trends in error rates, including Forward Error Correction rates, and adaptation may be used before packet error rates exceed an unacceptable level.

A system that investigates, identifies and cancels fraudulent transactions comprises a fraud detection server that receives a first dataset indicating a quantity of fraud-transactions. The first dataset is generated at least in part by a fraud-score model. The system receives a second dataset including a quantity of false positive fraud-transactions from the fraud transactions. The system calculates a fraud error rate using the quantity of fraud-transactions and the quantity of false positive fraud-transactions. The system generates a new fraud-score model when the fraud error rate exceeds a predefined error rate.

In a mobile communication system having channels with different bandwidths, a channel bandwidth can be dynamically selected for transmitting data over a connection between a radio access network and a mobile station. According to the invention, a channel bandwidth and also a target for a data error rate for the transmission is selected based on a determined transmission need, i.e. a determined characteristics of the data to be transmitted and on a determined load of a communication resource, e.g. channel power and / or total carrier power used in a cell. By making it possible to dynamically select a data error rate, a connection can, by increasing the data error rate and consequently decreasing the channel power, carry on using a certain channel bandwidth even if at a first instance a load threshold level is achieved for the channel power or the total power used in the cell. Thereby, the capacity of the system can be increased, while at the same time the throughput over the connection is increased.

Adjusting model parameters is described for a speech recognition system that combines recognition outputs from multiple speech recognition processes. Discriminative adjustments are made to model parameters of at least one acoustic model based on a joint discriminative criterion over multiple complementary acoustic models to lower recognition word error rate in the system.

The invention provides a self-adaptive transmission method of a large-scale MIMO downlink. The self-adaptive transmission method of the large-scale MIMO downlink comprises the following steps: firstly, utilizing a stronger correlation of a large-scale MIMO spatial channel to transform the spatial channel into a beam domain through a DFT transformation; in the beam domain, utilizing a maximum system and rate as a criterion to select uses communicate simultaneously, and assigning an optimal transmission beam for each user, thereby determining a transport layer number and a pre-coding matrix for each user; then, by taking inter-user interferences into consideration, calculating a MMSE balanced posterior signal to interference plus noise ratio on each resource element of each transport layer for each user; and then, acquiring an equivalent signal to interference plus noise ratio for each user by an equivalent signal to noise ratio mapping; finally, on the premise that a word error rate of the transmission is ensured, by each client, to be lower than a given value, determining and feeding back a channel quality indicator of each codon, aiming at maximizing the system throughput. By the self-adaptive transmission method of the large-scale MIMO downlink, the reliability of a large-scale MIMO wireless transmission is increased; and an excellent system spectral efficiency is acquired while the system word error rate is ensured to be lower than the given value.

A hypothesis space of a search graph may be determined. The hypothesis space may include n hypothesis-space transcriptions of an utterance, each selected from a search graph that includes t>n transcriptions of the utterance. An evidence space of the search graph may also be determined. The evidence space may include m evidence-space transcriptions of the utterance that are randomly selected from the search graph, where t>m. For each particular hypothesis-space transcription in the hypothesis space, an expected word error rate may be calculated by comparing the particular hypothesis-space transcription to each of the evidence-space transcriptions. Based on the expected word error rates, a lowest expected word error rate may be obtained, and the particular hypothesis-space transcription that is associated with the lowest expected word error rate may be provided.

A system and method for effectively implementing an optimized language model for speech recognition includes initial language models each created by combining source models according to selectable interpolation coefficients that define proportional relationships for combining the source models. A rescoring module iteratively utilizes the initial language models to process input development data for calculating word-error rates that each correspond to a different one of the initial language models. An optimized language model is then selected from the initial language models by identifying an optimal word-error rate from among the foregoing word-error rates. The speech recognizer may then utilize the optimized language model for effectively performing various speech recognition procedures.