Abstract: A quantizer includes: a clipper configured to clip a portion exceeding a quantization range of a sample value sampled at a predetermined rate; and a noise shaper configured to determine a plurality of candidates for a quantization level based on the clipped sample value, and outputs, as a quantization value, a value obtained by adding a minimum noise in which a noise in a low-frequency region is minimum among noises generated in each candidate to the sample value before clipping.

Abstract: According to an embodiment of the present invention, a picture decoding method performed by a decoding apparatus is provided. The method comprises: decoding image information comprising information on a quantization parameter (QP), deriving an expected average luma value of a current block from neighboring available samples, deriving a quantization parameter offset (QP offset) for deriving a luma quantization parameter (luma QP) based on the expected average luma value and the information on the QP, deriving the luma QP based on the QP offset, performing an inverse quantization for a quantization group comprising the current block based on the derived luma QP, generating residual samples for the current block based on the inverse quantization, generating prediction samples for the current block based on the image information and generating reconstructed samples for the current block based on the residual samples for the current block and the prediction samples for the current block.

Abstract: According to an embodiment of the present invention, a picture decoding method performed by a decoding apparatus is provided. The method comprises: decoding image information comprising information on a quantization parameter (QP), deriving an expected average luma value of a current block from neighboring available samples, deriving a quantization parameter offset (QP offset) for deriving a luma quantization parameter (luma QP) based on the expected average luma value and the information on the QP, deriving the luma QP based on the QP offset, performing an inverse quantization for a quantization group comprising the current block based on the derived luma QP, generating residual samples for the current block based on the inverse quantization, generating prediction samples for the current block based on the image information and generating reconstructed samples for the current block based on the residual samples for the current block and the prediction samples for the current block.

Abstract: In one embodiment, a computing system may access a first set of pixel values of an image having a bit depth. The first set of pixel values includes a first subset and a second subset. Each pixel value in the first subset is less than each pixel value in the second subset. The computing system may further determine an overflow range based on a pixel value overflow associated with the bit depth. The overflow range covers a second set of pixel values, which includes (1) the second subset and (2) an overflow representation of the first subset. In response to determining that the overflow range is less than a range covering the first set of pixel values, the computer system may then generate quantization levels corresponding to the overflow range and associate each pixel value of the second set of pixel values with one of the quantization levels.

Abstract: A method for encoding an input signal comprising signal frames into quantized bits is disclosed, the method comprises generating, for each frame of the input signal, a signal matrix comprising matrix coefficients obtained from that frame, grouping the matrix coefficients of each signal matrix into a plurality of partition vectors, and for each partition vector, selecting one vector quantization scheme from among a plurality of vector quantization schemes and quantizing that partition vector according to the selected vector quantization scheme to obtain the quantized bits. In an adaptive mode, the method comprises grouping differently the matrix coefficients obtained from different frames, and/or selecting different vector quantization schemes for partition vectors obtained from different frames.

Abstract: An audio processing device includes a feature extraction unit and signal generating unit. The feature extraction unit is configured to extract a feature quantity of a first audio signal for each of a plurality of periods. The signal generating unit is configured to for generate a second audio signal by time axis expanding/compressing either a section of the first audio signal in which the feature quantity is steadily maintained for a period time, or a section of the first audio signal in which a fluctuation of the feature quantity is repeated and excluding from the time axis expanding/compressing a section of the first audio signal in which a fluctuation of the feature quantity is not similar to that of other sections of the first audio signal.

Abstract: A method of signaling of coding modes including an IntraBC mode (Intra-block copy mode) to improve the coding efficiency is disclosed. The embodiment takes into consideration the facts that Inter prediction is more common in Inter slices than IntraBC prediction. In another embodiment, a binary sub-tree is disclosed that includes all the partition modes associated with IntraBC mode and the binary sub-tree consists of only said all members of the partition mode group.

Abstract: Embodiments relate to the field of PON technologies, and provide a method, apparatus, and system. The method includes: receiving capability information that is sent by an ONU by using a first preamble; calculating a tap coefficient based on the received first preamble if the capability information indicates that the ONU has an equalization capability; sending the tap coefficient to the ONU, where the ONU updates a tap coefficient of a first equalizer in the ONU based on the received tap coefficient, and when subsequently sending upstream data, sends, by using a second preamble, the upstream data equalized by using the first equalizer, and a length of the second preamble is less than a length of the first preamble; receiving the upstream data; and equalizing the received upstream data based on the second preamble by using a second equalizer in an OLT. In this way, upstream overheads can be reduced.

Abstract: An information processing method, an apparatus, a communications device, and a communications system are provided. The communications device is configured to: obtain a starting position in a buffer sequence W for an output bit sequence, and determine the output bit sequence from the buffer sequence W based on the starting position, where a value of the starting position is one element in {p0, p1, p2, . . . , pkmax?1}, 0?k<kmax, 0?pk<NCB, pk is an integer, k is an integer, NCB is a length of the buffer sequence W, kmax is an integer greater than or equal to 4, and there are two or more different neighboring intervals in {p0, p1, p2, . . . , pkmax?1}. A bit sequence for initial transmission or retransmission is properly determined, so that decoding performance of a communications device on a receive end after receiving the bit sequence is improved, a decoding success rate is enhanced, and a quantity of retransmission times is further reduced.

Abstract: A process of hiding one or more public keys inside of random noise is introduced, whose purpose is to protect the privacy of the public keys. In some embodiments, the random noise is produced by quantum randomness, using photonic emission with a light emitting diode. When the public key generation and random noise have the same probability distributions, and the key size is fixed, the security of the hiding can be made arbitrarily close to perfect secrecy, by increasing the noise size. The process of hiding can protect public keys that are vulnerable to Shor's algorithm or analogs of Shor's algorithm, executed by a quantum computer. The hiding process is practical in terms of infrastructure and cost, utilizing the existing TCP/IP infrastructure as a transmission medium, and a light emitting diode(s) in the random noise generator.

Abstract: A system uses multi-level encoding where each symbol of a plurality of symbols represents more than one bit of information in a user data symbol stream for transfer using a multilevel transmission channel. The user data symbols are represented in a digital bitwise form such that each symbol is presented as a plurality of bits and each bit is subject to a different probability of error. An error correction procedure is applied based on the different error probability that is associated with each bit in the plurality. The channel can be configured to support a mosaic tile structure, each tile containing a channel symbol such that a selected tile has a collective error probability that is different from other tiles. Customized coding can be applied to the tile structure to allocate a selected amount of error correction power to the selected tile based on an overall available correction power.

Abstract: The present technique relates to a frequency band extension apparatus, a frequency band extension method, and a program which are configured to more easily obtain a high quality sound signal. An input signal may be divided into sub-band signals of a plurality of sub-bands, powers of high frequency sub-bands of the input signal may be estimated based on feature values extracted from the input signal to obtain high frequency sub-band power estimation values, the high frequency sub-band powers obtained from the sub-band signals of high-frequency sub-bands of the input signal may be compared with the high frequency sub-band power estimation values, and a high-frequency signal of the input signal may be generated based on a result of the comparison and the sub-band signals.

Abstract: Methods and apparatus for enabling rapid transactions over a speed limited bus are disclosed. In one exemplary embodiment of the present disclosure, a host controller and an application specific integrated circuit (ASIC) are connected via an Inter-Integrated Circuit (I2C) Bus that is further adapted to enable a simplified signaling scheme. Unlike traditional I2C bus transactions which are flexible but speed limited, the simplified signaling scheme reduces bus overhead and enables rapid transactions. In an exemplary context, the simplified signaling scheme enables the ASIC to rapidly configure a series of photodiodes with different channel gain parameters so as to, for example, measure heartbeats by visually detecting a pulse within human flesh.

Abstract: In an RFID system, a method and apparatus for linking an RFID tag to an associated object. The system includes a relatively simple tag, a reader, a linker, and a store. The reader interrogates the tag for an ID and selectively provides the ID to the linker. The linker, in turn, uses the ID to provide back to the reader an associated Uniform Resource Identifier (“URI”). The reader then forwards the URI to the store. In response, the store returns to the reader the object associated with the ID via the URI. The disclosed method and apparatus provide more efficient and secure tag authentication.

Abstract: Devices and methods to compress sensor data are generally described herein. An exemplary wearable device to compress sensor data may include a sensor including circuitry to sense sensor data, and a communication circuit to receive, from a remote device, a detected link quality of a low-power communication channel used to communicate with the remote device. The communication circuit further to transmit compressed data to a remote device over the low power communication channel. The wearable device may further include a compressible sensor data module to apply a compression algorithm to compress received sensor data based on the detected link quality to provide the compressed sensor data.

Abstract: Disclosed are various embodiments directed to transcoding a bit stream comprising audio and video frames that contains discontinuities. The video and/or audio portion of the bit stream may be divided into segments, where some of the segments are transcoded in parallel using a plurality of transcoder devices. In some embodiments, a discontinuity is detected in the bit stream before encoding, where the discontinuity is associated with a shift in a set of time stamps associated with the bit stream. A magnitude of the discontinuity is determined. The transcoding system aligns at least a portion of the audio frames to the series of video frames according to at least the discontinuity and the magnitude of the discontinuity. In other embodiments, the discontinuity is detected after encoding by analyzing time stamps that have been established during the transcoding process.

Abstract: In one aspect, the invention provides an audio encoding method characterized by a decision being made as to whether the device which will decode the resulting bit stream should apply post filtering including attenuation of interharmonic noise. Hence, the decision whether to use the post filter, which is encoded in the bit stream, is taken separately from the decision as to the most suitable coding mode. In another aspect, there is provided an audio decoding method with a decoding step followed by a post-filtering step, including interharmonic noise attenuation, and being characterized in a step of disabling the post filter in accordance with post filtering information encoded in the bit stream signal. Such a method is well suited for mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Abstract: Disclosed is a method, apparatus and non-transitory computer readable storage medium, which is configured to examine an input audio signal to determine an estimated pitch period associated with a detection criterion, encode the filtered audio signal to generate a compressed audio bit stream, decode the compressed audio bit stream to generate a decoded audio signal and adaptively filtering the decoded audio signal to produce an output audio signal, wherein adaptively filtering the decoded audio signal comprises filtering each of a plurality of segments of the decoded audio signal in a manner that is dependent upon the estimated pitch period associated therewith, and wherein a pitch-based post-filter operates to shape noise components located between harmonics of the input audio signal.

Abstract: Methods of encoding and decoding for video data are describe in which significance maps are encoded and decoded using non-spatially-uniform partitioning of the map into parts, wherein the bit positions within each part are associated with a given context. Example partition sets and processes for selecting from amongst predetermined partition sets and communicating the selection to the decoder are described.

Abstract: A system uses multi-level encoding where each symbol of a plurality of symbols represents more than one bit of information in a user data symbol stream for transfer using a multilevel transmission channel. The user data symbols are represented in a digital bitwise form such that each symbol is presented as a plurality of bits and each bit is subject to a different probability of error. An error correction procedure is applied based on the different error probability that is associated with each bit in the plurality. The channel can be configured to support a mosaic tile structure, each tile containing a channel symbol such that a selected tile has a collective error probability that is different from other tiles. Customized coding can be applied to the tile structure to allocate a selected amount of error correction power to the selected tile based on an overall available correction power.

Abstract: A resistive memory device. Implementations may include an array of memory cells including resistive memory elements which are coupled to isolation transistors and which may include a magnetic tunnel junction. A decoder decodes input address information to select a row of the array. A binarizer coupled to the memory array assigns binary weights to outputs of the memory array output through bit lines coupled to the memory cells. A summer sums the binary weighted outputs, and a quantizer generates an output digital code corresponding to data stored in a plurality of memory cells during a prior program cycle. The outputs of the memory array may be currents or voltages. In implementations multiple arrays of memory cells may be utilized and their respective outputs combined to form higher bit outputs, such as eight bit, twelve bit, sixteen bit, and so forth.

Abstract: An input image is denoised by first constructing a pixel-wise noise variance map from the input image. The noise has spatially varying variances. The input image is partitioned into patches using the noise variance map. An intermediate image is determined from the patches. Collaborative filtering is applied to each patch in the intermediate image using the noise variance map to produce filtered patches. Then, the filtered patches are projected to an output image.

Abstract: Embodiments of methods and apparatus for reducing electromagnetic interference of a received signal are disclosed. One method includes receiving a signal over at least two conductors, extracting a common-mode signal from the at least two conductors, processing the common-mode signal, and reducing electromagnetic interference of the received signal by summing the processed common-mode signal with the received signal.

Abstract: A modulator and a method are disclosed. The modulator is for generating a band pass signal and comprises: sigma delta modulation logic operable to receive an input signal and to perform at least a 3-level quantisation of the input signal to generate an at least 3-level quantised signal; and requantisation logic operable to requantise the at least 3-level quantised signal to a 2-level quantised signal to be provided as the band pass signal. This approach improves the coding efficiency achieved compared to that possible using a 2-level sigma delta modulator, whilst also providing improved noise performance due to the inherent linearity of the 2-level quantised signal which is provided to drive the switch mode power amplifier. Accordingly, the performance of the modulator is improved by increasing its coding efficiency whilst maintaining its linearity which improves the noise performance in adjacent channels.

Abstract: A method is disclosed for compressing a sequence of initial digital values into a compressed sequence of compressed values, intending to restore these values into a decompressed sequence of decompressed values. For a first initial value of the sequence, the compressed value of the first initial value is equal to the first initial value and the decompressed value of the compressed value of the first initial value is equal to the first initial value.

Abstract: One embodiment includes an adaptive sample quantization system. The adaptive sample quantization system includes an antenna to receive a radio frequency (RF) signal having data encoded therein, and analog antenna electronics configured to convert the RF signal to an analog electrical signal. The system also includes an analog-to-digital converter (ADC) directly coupled to the antenna and configured to generate a plurality of consecutive digital samples of the RF signal. The system further includes a quantizer to determine a mode based on the plurality of consecutive digital samples and to select at least one threshold based on the determined mode. The quantizer can further compare each digital sample with the at least one threshold to generate a corresponding one of a plurality of output samples having a reduced number of bits relative to the respective digital sample to substantially mitigate potential interference and facilitate extraction of the data.

Abstract: A control method for an encoding apparatus which performs bit rate control includes an acquiring step of acquiring a first quantization parameter as a quantization parameter to be used for quantizing a rate control unit, a first calculating step of calculating an evaluation value from a preset rate and a rate of an actually encoded rate control unit, a first setting step of, if the evaluation value is within a predetermined range, setting the first quantization parameter acquired by the acquiring step as a quantization parameter to be used for quantizing a rate control unit, and a second setting step of, if the evaluation value is not within the predetermined range, setting a second quantization parameter on the basis of the calculated evaluation value as a quantization parameter to be used for quantizing a rate control unit.

Abstract: On a criterion to minimize the entropy of the linear prediction residual of the input signal used for calculation of the input PARCOR coefficient sequence, PARCOR coefficients with larger absolute values are quantized with higher quantization precisions so as to reduce the increase of the code amount of the linear prediction residual caused by the quantization error of the PARCOR coefficients. If the PARCOR coefficient is represented by a value formed by a predetermined number of bits, the number of effective bits from the most significant bit toward the least significant bit included in the output value increases with the absolute value of the PARCOR coefficient.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: The transmission power is decreased in an appropriate manner, while suppressing the deterioration of the line quality to the minimum. A mobile communication terminal 1 controls the transmission power in response to a request given from a base transceiver station 2. When an average transmission power Wa that is an average value of the transmission power in a monitoring period Ta is smaller than a threshold value TH, mobile communication terminal 1 continuously controls the transmission power in response to the request given by the base transceiver station 2. When the average transmission power Wa in the monitoring period Ta is equal to or greater than the threshold value TH, the transmission power is controlled in a transmission power suppressing period Tb that follows the monitoring period Ta so that the average value of the transmission power becomes equal to or smaller than the threshold value in a control period Tc including the monitoring period Ta and the transmission power suppressing period Tb.

Abstract: A method and apparatus for processing feedback implemented in a wireless transmit/receive unit (WTRU) comprises estimating a channel matrix. The effective channel is calculated and a precoding matrix is selected. Feedback bits are generated and transmitted.

Abstract: A video processor is described, which is useful for implementing a quantization process, in compliance with the H.264 standard. The video processor includes an input, for receiving a block of image data. The image data is loaded into an internal register. In response to receiving a SIMD instruction, a quantizer, which incorporates the quantization lookup tables associated with the H.264 standard in its associated hardware, makes necessary high-level quantization decisions. In response to receiving another SIMD instruction, the quantizer uses those high-level quantization decisions to retrieve specific values from the quantization lookup tables.

Abstract: An analog to digital converter with increased sub-range resolution. The device includes an analog front end that produces an analog communication signal, a digital front end that receives a digital communication signal, and an Analog to Digital Converter (ADC) that samples the analog communication signal across a full-range. The ADC includes a full-range ADC having a first quantization accuracy configured to sample the analog communication signal across the full-range and a central sub-range ADC having a second quantization accuracy greater than the first quantization accuracy and configured to sample the analog communication signal across a central sub-range of the full-range. The ADC also includes signal combining circuitry configured to process outputs of the full-range ADC and the central sub-range ADC to create the digital communication signal.

Abstract: A method, a system, and a service control point (SCP) for realizing call management are provided. The method for realizing call management includes: inquiring user information of a calling terminal according to a call request message of the calling terminal; sending the user information of the calling terminal, over a called terminal; and exercising a corresponding call control according to a response made by the called terminal to a call carrying the user information of the calling terminal. With the method, system, and SCP for realizing call management provided by the embodiments of the present invention, the called terminal may selectively answer the call according to the user information of the calling terminal, thereby avoiding disturbing the work due to answering some unnecessary calls.

Abstract: Provided is a wireless communication device which can improve the notification accuracy of the channel state information (CSI) without straining the feedback circuit. In this device, a channel estimator (105) uses a pilot signal input from a wireless receiver (102) to estimate the channel and obtain a plurality of path gains in each of a plurality of delay periods. Then, the channel estimator (105) outputs the plurality of path gains to a quantizer (107). The quantizer (107) quantizes the plurality of path gains in the number of notification bits corresponding to each of the plurality of delay periods based on the correspondence of the delay period and the number of notification bits input from a setting unit (106).

Abstract: A data modulation circuit has an adder adding an input signal, and an output signal of a memory device; and an output circuit part discriminating and quantizing the output signal of the adder by a predetermined threshold value. The memory device receives and holds the output signal of the adder and a predetermined signal, and supplies the held signals to the adder as an output signal of the memory device.

Abstract: A method applies a parametric approach to bandwidth extension but does not require training. The method computes narrowband linear predictive coefficients from a received narrowband speech signal, computes narrowband partial correlation coefficients using recursion, computes Mnb area coefficients from the partial correlation coefficient, and extracts Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors. The method further comprises synthesizing a wideband signal using the wideband LPCs and a wideband excitation signal, highpass filtering the synthesized wideband signal to produce a highband signal, and combining the highband signal with the original narrowband signal to generate a wideband signal.

Abstract: A method and apparatus reliably encode and decode information over a communication system. The method includes transforming two coefficients into two pairs of random variables, one random variable in each pair having substantially equal energy as one random variable in the other pair. The method further includes quantizing each of the pairs of random variables and entropy coding each quantized random variable separately creating an encoded bitstreams. The encoded bitstreams are received by a decoder which first determines which channels of the communication system are working. The encoded bitstream is entropy decoded, inversed quantized and inversed transformed. An inverse transform performs three different transformations depending upon which channels are working, i.e., whether the first, second or both channels are working.

Abstract: The digital delta-sigma modulator includes a signal input for receiving digital samples of N bits, and a digital filter connected to the signal input. The digital filter performs addition/subtraction and integration operations according to a redundant arithmetic coding for delivering digital filtered samples. A quantizer performs a nonexact quantization operation so as to deliver digital output samples of n bits, with n being less than N. The input of the quantizer is connected within the digital filter.

Abstract: Encoding and decoding methods that perform quantization using adaptive reconstruction levels are presented. The reconstruction levels for a given partitioning of the data space may be selected based upon the distribution of data points within each sub-part defined by the partitioning. In some cases, the adaptive reconstruction levels may be based upon an average of the data points within each sub-part. In some cases, the adaptive reconstruction levels may be selected using a rate-distortion analysis including the quantization distortion associated with the levels versus the data points and the rate associated with transmitting the encoded adaptive reconstruction levels. The methods relate to data compression in a range of applications including audio, images and video.

Abstract: Certain aspects of the present disclosure relate to a method for emulating N-bits uniform quantization of a received pulse signal by using one-bit signal measurements. One method for wireless communications includes receiving a signal transmitted over a wireless channel, wherein the signal comprises a sequence of pulses, applying gain values to the pulses to obtain a binary matrix for each gain value, generating a probability vector for each gain value using the binary matrix obtained for that gain value, and generating an output signal by linearly combining the probability vectors.

Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.

Abstract: A data element can be encoded into multiple encoded data elements using an encoding algorithm that includes an encoding function and one or more encoder constant. The encoded data elements can be organized into multiple pillars, each having a respective pillar number. Each of the pillars is sent to a different storage unit of a distributed storage network. To recover the original data element, the encoded data elements are retrieved from storage, and the encoder constant is recovered using multiple encoded data elements. Recovering the encoder constant allows the encoding algorithm originally used to encode the data elements to be determined, and used to recover the original data element. The security of the stored data is enhanced, because an encoded data element from a single pillar is insufficient to identify the encoder constant.

Abstract: There are provided a transmission and reception device having a function for correcting a data error in a communication path. In the transmission device, a redundant bit addition unit adds a redundant bit to each data bit which has been divided by one bit by a division unit; and an interleaver performs interleave. The transmission device transmits a signal which has been subjected to FM modulation by an FM modulation unit. In the reception device, a symbol decision unit performs a symbol decision at a Nyquist point for a signal which has been FM-demodulated by an FM demodulation unit; a bit conversion unit performs bit conversion according to the result of symbol decision; and a frame recovery unit deletes the redundant bit added by the redundant bit addition unit of the transmission device, from the bit string de-interleaved by a de-interleaver. Thus, it is possible to surely perform an error correction with a simple configuration even when the communication state is not in a preferable environment.

Abstract: Techniques, apparatuses and systems for providing communications based on time reversal of a channel impulse response of a pulse in a transmission channel between a transmitter and a receiver to enhance reception and detection of a pulse at the receiver against various effects that can adversely affect and complicate the reception and detection of the pulse at the receiver.

Abstract: There are provided a transmission and reception device having a function for correcting a data error in a communication path. In the transmission device, a redundant bit addition unit adds a redundant bit to each data bit which has been divided by one bit by a division unit; and an interleaver performs interleave. The transmission device transmits a signal which has been subjected to FM modulation by an FM modulation unit. In the reception device, a symbol decision unit performs a symbol decision at a Nyquist point for a signal which has been FM-demodulated by an FM demodulation unit; a bit conversion unit performs bit conversion according to the result of symbol decision; and a frame recovery unit deletes the redundant bit added by the redundant bit addition unit of the transmission device, from the bit string de-interleaved by a de-interleaver. Thus, it is possible to surely perform an error correction with a simple configuration even when the communication state is not in a preferable environment.

Abstract: Methods and systems for signal quantization may include, but are not limited to: receiving a complex signal characterized by a real portion and an imaginary portion; computing a vector magnitude of a sample of the complex signal; comparing the vector magnitude of the sample to a quantization threshold value; and associating a quantized magnitude with the sample according to a comparison between the vector magnitude of the sample and the quantization threshold value.

Abstract: Certain embodiments of the present disclosure allow a receiver to utilize both a minimum mean square error (MMSE) detection algorithm and a maximum likelihood (ML) detection algorithm. The receiver may control, based on various conditions, a mode selection signal to select between output generated with the MMSE detection algorithm and the ML detection algorithm. By sharing logical components, certain embodiments may allow a receiver to implement both ML and MMSE detection algorithms with minimal computational overhead when compared to implementing a single one of the decoding algorithms.

Abstract: An analog to digital converter with increased sub-range resolution and method for using the analog to digital converter is described herein. The device includes an analog front end that produces an analog communication signal, a digital front end that receives a digital communication signal, and an Analog to Digital Converter (ADC) that samples the analog communication signal across a full-range. The ADC includes a plurality of sub-range ADCs, each sub-range ADC measuring the analog communication signal across at least one respective sub-range of the full-range, the plurality of sub-ranges extending across the full-range, a central sub-range ADC having greater quantization accuracy than at least one other sub-range ADC. The ADC also includes signal combining circuitry operable to process outputs of the plurality of sub-range ADCs to create the digital communication signal.