Abstract: Provided is an audio-signal dynamic-range-compression system. An input line accepts an input signal. A filter set coupled to the input line divides the input signal into a set of frequency bands, including a first frequency band and a second frequency band. A compressor has a first input that receives the first frequency band, a second input and an output. An adder has a first input coupled to the output of the compressor, a second input coupled to the second frequency band, and an output. An overshoot estimation module has an input coupled to the output of the adder and an output coupled to the second input of the compressor. The output of the overshoot estimation module provides an overshoot signal that indicates a result of a comparison between a signal at the input of the overshoot estimation module and a specified threshold.

Abstract: A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

Abstract: Embodiments are directed towards modifying a quality of an image without affecting a bit rate associated with a video sequence of images. For each macro-block (MB) or sub-block within a MB for a target image to be encoded, various statistics are determined for a luminance component and chrominance components that provide at least average values for the chrominance components that may then be used to identify saturation thresholds. The saturation thresholds are used to determine a Qlevel. An average Qlevel from at least a previous frame is used to generate a Q modulation factor that may be combined with an activity based modulation factor and/or variance based modulation factor, or used singly. The final quantizer is calculated by bit-rate controller base quantizer multiplied by the Q modulation factor, and may be used to encode a MB or sub-block within a MB.

Abstract: Systems and method for managing and/or mitigating the impact of bit errors on encoded frames received by an LC-SBC (Low Complexity Sub-band Coding) decoder are described herein. For example, in one embodiment, the impact of bit errors on an LC-SBC frame received by an LC-SBC decoder is estimated and one of a plurality of bit error management techniques is applied to the LC-SBC frame based on the estimated impact, wherein the bit error management techniques may include performing PLC, performing normal SBC decoding, or performing some other technique for managing and/or mitigating the impact of the bit errors. Techniques for concealing bit errors in LC-SBC frames are also described.

Abstract: The derivative of an input signal is level-crossing-sampled and the resulting samples are transmitted. At the receiver, the resulting samples are fed to a zero-order hold followed by an integrator, which results in piecewise-linear reconstruction. The systems and methods are further refined to reduce the number of samples generated per unit of time, compared to methods based on zero-order-hold reconstruction, for a given signal-to-error ratio, without significant hardware overhead.

Abstract: This invention is for a method for transforming a seismic trace into a compressed domain. The seismic source wavelet is transformed into a zero degree phase wavelet and a shifted 90 degree phase wavelet, and the two wavelets span a 2-dimensional sub-space. A dictionary is created by collecting the wavelets in the sub-space. In practice this dictionary is usually combined with conventional existing wavelet dictionaries. The seismic trace is projected onto the dictionary (sub-space alone or combined) to find the best matching projection, with a residual determined after each projection, wherein the sum of the residuals determines the fidelity of the data compression.

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: A device is provided for use of an antennal base formed of two antennas which pick up the emissions present and produce two radioelectric signals S1 and S2. These two signals are used to produce at least one intermediate-frequency signal Fl by demodulation of one of the two signals by the other (autotransposition). The demodulation is carried out by firstly transposing one of the signals, S1 for example, around a given frequency F1, the signal S2 being preserved around its initial central frequency F0. Thus, whatever the central frequency F0 of the emission picked up by the antennas, the demodulation produces a signal of central frequency F1, thereafter demodulated into a given intermediate frequency Fl by a local oscillator of constant frequency F2=F1+Fl. The device is applied to the production of a device for detecting emissions and for characterizing the emissions picked up.

Abstract: An information handling system device includes a plurality of electronic components; an electric circuit including at least one trace for connecting two or more of the plurality of electronic components and transmitting data between the plurality of electronic components via at least one electric signal; and a substrate including an insulating material for serving as a base for the electric circuit, wherein each of the at least one electric signal transmitted between the plurality of electronic components is transmitted utilizing slope manipulation by manipulating each of the at least one electric signal to provide a slope substantially proportional to a discrete integer data value of n discrete integer data values, n being a positive integer greater than or equal to 3, said discrete integer data value represented by using one of n distinct slopes transmitted utilizing a particular reference voltage of n predetermined reference voltages.

Abstract: A wireless transmission device of the present invention includes n (where n is an integer of two or more) transmission antennas and a delay imparting section for delaying transmission signals supplied to the n transmission antennas by a maximum delay time (n?1)T or less based on a delay time T dependent upon a communication signal, which indicates whether to transmit the transmission signals by way of frequency diversity or multiuser diversity.

Abstract: A method and apparatus for detecting a radar signal is provided. The method includes collecting a plurality of pulses based on a reference time or a reference pulse collection count, classifying the plurality of pulses into groups based on similarity of pulse widths, classifying the groups into a short-pulse type and a long-pulse type and calculate the number of groups in each type, and determining whether a radar signal exists based on a number of groups in each type.

Abstract: A precision charge dump circuit configured to transfer preset quanta of charge to or from a first capacitor (for example, an integration capacitor in an in-pixel ADC circuit). In one example, the charge dump circuit uses a second capacitor that is pre-charged with the preset quanta of charge to determine the preset value of the quanta of charge, and an amplifier in a voltage-follower mode to cause the charge subtraction or addition.

Abstract: An information handling system device includes a plurality of electronic components; an electric circuit including at least one trace for connecting two or more of the plurality of electronic components and transmitting data between the plurality of electronic components via at least one electric signal; and a substrate including an insulating material for serving as a base for the electric circuit and the plurality of electronic components, wherein the at least one electric signal transmitted between the plurality of electronic components is transmitted utilizing slope manipulation to provide a slope directly proportional to a data value.

Abstract: A motion estimation method, medium, and system with fast motion estimation. The motion estimation method includes comparing a cost indicating a difference between a current block of a current image and a block of a reference image specified by a starting point with a predetermined threshold and selectively searching for the best matching block of the current block from the starting point according to the comparison result.

Abstract: A method and apparatus for evaluating and optimizing a signaling system is described. A pattern of test information is generated in a transmit circuit of the system and is transmitted to a receive circuit. A similar pattern of information is generated in the receive circuit and used as a reference. The receive circuit compares the patterns. Any differences between the patterns are observable. In one embodiment, a linear feedback shift register (LFSR) is implemented to produce patterns. An embodiment of the present disclosure may be practiced with various types of signaling systems, including those with single-ended signals and those with differential signals. An embodiment of the present disclosure may be applied to systems communicating a single bit of information on a single conductor at a given time and to systems communicating multiple bits of information on a single conductor simultaneously.

Abstract: An apparatus, system, method and computer program product for secure ranging between at least two devices in radio communications with each other. In particular, an apparatus, system, method and computer program product for secure ranging between at least two devices communicating via ultra wideband (UWB) protocols. Either ternary-IR or time-hope-IR sequences are used for ranging and security. A first device transmits a range packet. A second device responds to the range packet after a delay time known only to the first and second devices. The delay time can be selected randomly by either to the first device or the second device and made known to the other device in an encrypted notification packet.

Abstract: At a base station apparatus, known pilot signals for use in channel estimation are transmitted, and in addition thereto MCS pilot signals that are used to perform adaptive modulations respectively corresponding to a plurality of modulation schemes are multiplexed and output. A mobile station apparatus dispreads the respective MCS signals out of the multiplex signal, compares these to known symbols patterns, and sends the MCS pilot signals that show a matching relationship to the base station apparatus as a mobile station reception result. Upon receiving the mobile station reception result from the mobile station apparatus, the base station apparatus selects the modulation scheme of the optimum modulation level for the downlink signals. This configuration makes it possible to switch the modulation schemes in an accurate and simple way.

Abstract: Methods and apparatus for transmitting and receiving symbols in a communication system having a plurality of frequency bands are provided. A first symbol is transmitted on a first tone using a first constellation mapping and a second symbol is transmitted on a second tone using a second constellation mapping different from the first constellation mapping. The first symbol is retransmitted on the second tone using the second constellation mapping and the second symbol is retransmitted on the first tone using the first constellation mapping. The first and second symbols are detected from a combination of the received transmitted and retransmitted first and second signals.

Abstract: A processor-implemented method of estimating the saturation level of a transmitter from a transmitted communication signal received from the transmitter comprises generating a received data signal from the transmitted communication signal received from the transmitter, generating a transmitted data signal from the received data signal that corresponds to the actual data signal generated at the transmitter, comparing the transmitted data signal to the received data signal, and estimating the saturation level of the transmitter based on the comparison of the transmitted data signal to the received data signal.

Abstract: Coded digital data symbols sent from a transmitter through a transmission channel of a communications network are received in a receiver. An estimate, represented by a first number of bits, of a sent data symbol is calculated, and a second number of bits, lower than the first number, is selected from the estimate to achieve a rounded estimate represented by the second number of bits. The rounded estimate is decoded to achieve a decoded data symbol. A target value for a block error rate of the transmission channel is received from the network; and the second number of bits is selected in dependence on the target block error rate value. Thus an optimal rounded estimate is provided in most situations, and the method can be performed with the limited computational resources of a terminal.

Abstract: An apparatus and method for estimation of signal-to-noise ratio (SNR) with low bias that is effective for both positive SNRs and small to negative SNRs. The estimation is based on an iterative solution for the maximum likelihood estimate of the amplitude from which the SNR can be computed. The estimation is applicable for various modulated systems, including BPSK, QPSK and MPSK.

Abstract: This invention produces data packets that can vary in length and/or data compression ratio. First, an algorithm is employed to transform a data signal into fixed or variable length data packets at variable data compression ratios. If the algorithm produces fixed length data packets, the fixed length data packets are then converted to variable length data packets, which include only the valid data bytes of the fixed length data packets. Finally, the variable compression ratio, variable length data packets are provided with length codes at each end of each data packet to facilitate bi-directional searching and decompression. The transition from fixed to variable length data packets employs a buffer which stores the fixed length data packets until the fixed length data packets are converted to variable length data packets.

Abstract: A decoding system to achieve rates higher than 33.6 kbps in the analog modem to digital modem direction. The decoding system modifies the standard Tomlinson Harashima Precoding algorithm to adapt it for use in PCM modems. Instead of an arithmetic modulo operation that is implemented in the transmitter, the invention defines a Discrete Modulo Operation that performs the function of limiting the amplitude of the transmitted signal.

Abstract: An apparatus (200) checks voice quality when in a vocoder bypass mode. If the voice quality level is beyond an acceptable range, the apparatus (200) ends the vocoder bypass mode and returns to tandem vocoding. For example, a vocoder bypass controller (206) includes a speech quality detector (225) that determines the speech quality level of speech information based on at least one of detected volume level, echo level and noise level of the speech information. The vocoder bypass controller (206) with the speech quality detector (225) outputs a speech quality-based vocoder bypass control signal (236) to selectively activate or deactivate a vocoder bypass operation in response to the speech quality-based vocoder bypass control signal (236). In another embodiment, a network element for communicating speech packets includes an incoming decoder (512) and a speech quality detector (514) operative to determine the speech quality level of incoming speech packets.

Abstract: A technique for identifying the encoding law utilized by a central office codec may be implemented in a receive modem. The encoding law, which is typically dictated by the country in which the central office is located, is employed to generate a plurality of transmission levels during an initialization period associated with the modem system. The receive modem analyzes a number of these transmission levels to determine whether the levels have certain characteristics associated with the particular encoding law followed by the central office codec. When the receive modem detects the codec type, it may transmit a suitable identifier back to the transmit modem.

Abstract: A subscriber cluster unit for a wireless telecommunication system provides a wireless interface with a base station for a plurality of subscriber units. The cluster unit has a plurality of frequency agile modems for processing wireless communications with the base station and a plurality of subscriber line circuits, each for providing a telecommunication connection with a subscriber unit. A control processor assigns a modem for each communication between the base station and a selected subscriber unit which is coupled to one of the line circuits and associates that line circuit with the assigned modem for that communication. Thus, a subscriber unit coupled with any of the line circuits can communicate with the base station via any of the modems.

Abstract: A data conversion device converts a series of digital signals into an analog signal. The device has a current selection circuit that can vary the direction and magnitude of the current it outputs and an integrating circuit that integrates the current supplied from the current selection circuit to output a voltage. The direction of the current is determined in accordance with the digital signals fed in. When the absolute value of the current is great, the value by which the absolute value of the current is increased is decreased, and the value by which the absolute value of the current is decreased is increased.

Abstract: A data communication system and method that reduces conversion distortion on data calls between locations that encode telephone signals according to different conversion standards is disclosed. The data communication system (40) includes a first switch (44) and a processor (50). The first switch (44) receives a telephone call carrying a first analog modulated data signal, and converts the first analog modulated data signal to a first digitized data signal according to a first conversion standard. Next, the processor (50) converts the first digitized data signal back to the first analog modulated data signal and then demodulates the first analog modulated data signal to recover the original data signal.

Abstract: A technique for identifying the encoding law utilized by a central office codec may be implemented in a receive modem. The encoding law, which is typically dictated by the country in which the central office is located, is employed to generate a plurality of transmission levels during an initialization period associated with the modem system. The receive modem analyzes a number of these transmission levels to determine whether the levels have certain characteristics associated with the particular encoding law followed by the central office codec. When the receive modem detects the codec type, it may transmit a suitable identifier back to the transmit modem.

Abstract: A telephone apparatus of a telephone system, including at least one server for connecting a plurality of telephone apparatuses connected to a computer network, comprises: a communication circuit means for processing data to and from said computer network; an audio input means for generating a digital audio signal from input voice; an audio compression means for compressing the audio signal from the audio input means and for supplying the compressed audio signal to the communication circuit means; and a control means for determining a compression rate of the audio compression means in response to information on the ratio of use of the computer network obtained from the server.

Abstract: A method is presented to enable the impulse response of a channel to be identified when the received signal contains a large amount of quantization noise. A probabilistic estimate of the distribution of the quantization noise is subtracted from the summation of the products of the received training signal samples and tap coefficients to provide a more accurate error signal for the adjustment of the tap coefficients.

Abstract: Disclosed is a system for selectively enabling intermediate data processing of digital signals external to an integrated circuit (IC). The system comprises a transcoder, a codec connected to the transcoder, and a data processor located externally to the IC. In response to a strobe signal, a programmable switch diverts digital signals between the transcoder and the codec to a data path from the IC to the external processor. The digital signals are formatted within the IC for processing by the external processor. The externally processed digital signals are returned to the IC via a data path from the external processor. The digital signals are then reformatted within the IC for further processing by the IC.

Abstract: Apparatus for determining a complexity measure of a data signal is disclosed which includes an encoder, having an input terminal responsive to a data input signal and a data output terminal producing a coded output signal representing the data input signal at a constant bit rate. The encoder includes a variable quantizer, responsive to the data input signal, for producing a quantized signal, representing the data input signal, having a quantizing step size defined in response to a quantizing step size control signal. A bit rate regulator produces the quantizing step size control signal in response to the bit rate of the coded output signal and a quota input signal. A complexity analyzer generates a complexity representative signal related to the quantizing step size and the bit rate of the coded output signal.

Abstract: A speech encoder converts analog speech signals into a stream of binary coded speech samples. A circuit for performing analog to digital conversion compares an input speech signal with a signal from a digital integrator. Based on the comparison a series of decisions are registered which indicate a step sign of the input signal. A syllabic filter generates a step magnitude according to the registered decisions. The step sign and step magnitude are combined providing a sign/magnitude representation of the input signal. When the sign/magnitude representation is integrated digitally in a decimation filter a digital representation of the analog signal is formed. A speech decoder, the antithesis of the encoder, converts binary coded speech samples into an analog waveform in a manner opposite to the A/D conversion method.

Abstract: Disclosed is a system for selectively enabling intermediate data processing of digital signals external to an integrated circuit (IC). The system comprises a transcoder, a codec connected to the transcoder, and a data processor located externally to the IC. In response to a strobe signal, a programmable switch diverts digital signals between the transcoder and the codec to a data path from the IC to the external processor. The digital signals are formatted within the IC for processing by the external processor. The externally processed digital signals are returned to the IC via a data path from the external processor. The digital signals are then reformatted within the IC for further processing by the IC.

Abstract: A digital telephone (60) comprises send equipment (40) and receive equipment (50), coupled via a signal splitter/combiner (61) onto a digital telephone line (62) for communication with a remote party (90). The send equipment (40) comprises an A-law map (12) and a mu-law map (43). Receive equipment (50) comprises an inverse A-law map (14) and an inverse mu-law map (52). Maps (12,43) are selectable by the user (80). Similarly, inverse maps (14,52) are also selectable by the user (80). Each of send equipment (40) and receive equipment (50) may further comprise an un-quick-fix module (45) for removing unwanted insertions of quick fixes (25) inserted into the telephone line (62) by international telephone companies. Receive equipment (50) may further comprise a digital signal processor (57), computer (56), a set of prestored templates (58) corresponding to possible incoming telephone characteristics, and a caller parameter decipher module (51).

Abstract: In a terminal adapter which is located between a data terminal equipment unit (11) and a data communication network (12) and which includes a buffer memory (22, 27) for memorizing an original data signal sequence sent from each of the data terminal equipment unit and the data communication network, the original data signal sequence is subjected to data compression by a data compression circuit to be thereafter memorized into the buffer memory and to be read out of the buffer memory as a readout data signal sequence. The readout data signal sequence is expanded by a data expansion circuit (32, 35) to be sent to the data terminal equipment unit or the data communication network at a transmission rate matched therewith. The data compression may be executed by the use of an LZ method, a modified LZ method, such as LZW, LZSS, or the like.

Abstract: A speech coding transmission system is disclosed in which a coder of a differential coding system quantizes a difference between a predictive value from an adaptive predictor and an input speech signal by an adaptive quantizer in accordance with the level of the latter and to send out the resulting coded signal to transmission medium, and a decoder outputs a reproduced speech signal on the basis of a residual signal obtained by inversely quantizing a received coded signal with an inverse adaptive quantizer and a predictive value from an adaptive predictor. Both or either one of the coder and the decoder includes a comparator-attenuator which attenuates the amplitude of the input signal or reproduced speech signal to a value smaller than a predetermined threshold value and applies to the amplitude-attenuated speech signal to the adaptive predictor.