Abstract: Methods and systems are presented for characterizing a printer, display or other color reproduction device in which parametric and nonparametric forward color transforms are generated to construct a forward color device transform to characterize the mapping of CMYK input data to La*b* data of the device, with the parametric transform adapted to compensate for drifting performance of the color reproduction device.

Abstract: Methods are described for configuring a quantizer to achieve improved end-to-end distortion performance when transmitting encoded source data over a noisy channel. The codebook and partitioning are selected using an iterative process of determining an updated codebook and an updated partition space, where the updated codebook is based, in part, on the average symbol error probability of the channel. Complete knowledge of the transitional probabilities of the channel is not required. Variants of the iterative process are described.

Abstract: A CODEC circuit for POTS system comprises a digital circuit for outputting a digital control signal having a first bit width, the digital control signal being indicative of a voltage to be applied to a POTS subscriber line pair. Further, the CODEC circuit comprises a noise shaper coupled to an output of the digital circuit for generating a noise-shaped control signal and a digital-to-analog converter coupled to an output of the noise shaper, the input of the digital-to-analog converter having a second bit width being larger than 1 and smaller than the first bit width.

Abstract: Systems and methods for adaptive signal sampling and sample quantization for resource-constrained stream processing. Exemplary embodiments include a signal sampling and signal quantization method for a data stream from a ECG sensor into a pervasive device, the method including adaptively sampling the data stream, compressing data samples from the data stream, reducing the number of samples via quantization, defining a utility function for signal reconstruction, and formulating optimization on the reconstructed signal that is jointly sampled and quantized.

Abstract: The present invention provides an OOB detection circuit capable of making accurate signal determination even in the case where a characteristic fluctuation occurs in an analog circuit, thereby preventing deterioration in the yield of a product. To an amplitude determining circuit, a characteristic adjustment register for changing setting of an amplitude threshold adjustment mechanism for distinguishing a burst and a squelch from each other provided for the amplitude determining circuit is coupled. The characteristic adjustment register is controlled by a self determination circuit. An output of the amplitude determination circuit is supplied to a time determining circuit and also to the self determination circuit. On the basis of the output of the amplitude determining circuit, the self determination circuit controls the characteristic adjustment register.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

Abstract: Methods are described for configuring a quantizer to achieve improved end-to-end distortion performance when transmitting encoded source data over a noisy channel. The codebook and partitioning are selected using an iterative process of determining an updated codebook and an updated partition space, where the updated codebook is based, in part, on the average symbol error probability of the channel. Complete knowledge of the transitional probabilities of the channel is not required. Variants of the iterative process are described.

Abstract: Disclosed is a wideband audio signal coding/decoding device and method that may code a wideband audio signal while maintaining a low bit rate. The wideband audio signal coding device includes an enhancement layer that extracts a first spectrum parameter from an inputted wideband signal having a first bandwidth, quantizes the extracted first spectrum parameter, and converts the extracted first spectrum parameter into a second spectrum parameter; and a coding unit that extracts a narrowband signal from the inputted wideband signal and codes the narrowband signal based on the second spectrum parameter provided from the enhancement layer, wherein the narrowband signal has a second bandwidth smaller than the first bandwidth. The wideband audio signal coding/decoding device and method may code a wideband audio signal while maintaining a low bit rate.

Abstract: A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.

Abstract: An apparatus and method for processing an encoded signal are discussed. The method includes: if a coding type of an audio signal is a mixed signal coding type, extracting spectral data and a linear predictive coefficient from the audio signal; generating a residual signal for the linear prediction by performing an inverse frequency conversion on the spectral data; reconstructing the audio signal by performing a linear prediction coding on the linear predictive coefficient and the residual signal; and reconstructing a high frequency region signal using an extension base signal corresponding to a partial region of the reconstructed audio signal and band extension information.

Abstract: In a particular embodiment, a circuit device is disclosed that includes a data generator adapted to output a random pulse sequence having a particular spectral shape. The data generator includes a feedback loop with a transfer function, the output of which may be altered by limiter circuitry to increase stability of the data generator. The circuit device may further include, for example, a pulse edge control circuit to selectively apply a carrier suppression operation to at least one pulse-width modulated (PWM) signal in response to the random pulse sequence to produce at least one modulated PWM output signal. In such an example, the spectral energy associated with a PWM carrier of the modulated PWM output signal at a carrier frequency and associated harmonics is changed such that the modulated PWM output signal has a spectral shape defined by the particular spectral shape.

Abstract: An analog/digital gain control device avoid some of the requirements associated with the nature of a closed-loop AGC circuits and which meets the remaining requirements without much difficulty uses an analog to digital conversion method that increases the number of effective ADC bits by compressing the baseband input analog signal using a logarithmic circuit. After the compressed analog signal is converted into a digital signal, a digital anti-log process or look-up table (LUT) is used to expand the digital signal back to the original linear scale. The word size of the output of the anti-log process is larger than the input word size due to the nature of the anti-log function. To reduce the word size of the digital signal an open loop normalization technique can be applied.

Abstract: According to the present invention, a shift amount Sj?1 of the previous frame is held in a shift amount buffer of an integer signal coder. At least as many last sample values in the previous frame as the number of an order P used in linear predictive analysis are held in a sample buffer of the integer signal coder. The last P sample values in the previous frame held in the sample buffer of the integer signal coder are corrected in an interframe correction section by Sj-Sj?1 on the basis of the shift amount Sj of the current frame and the shift amount Sj?1 of the previous frame determined by a shift amount determining section.

Abstract: An encoding method includes partitioning an input stream into first sub-blocks and second sub-blocks, encoding the second sub-blocks based on a number and location of reference sub-blocks to generate third sub-blocks when at least one of the first sub-blocks corresponds to a forbidden pattern, and rearranging the first sub-blocks and the third sub-blocks to generate an output stream. The reference sub-blocks indicate the second sub-blocks that correspond to a reference pattern and the third sub-blocks includes a redundant information sub-block and non-redundant information sub-blocks. Thus, the encoding method may properly control a direct current (DC) component.

Abstract: A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.

Abstract: Methods and devices are used for switching between sound signal coding modes and for producing from a decoded target signal, an overlap-add target signal in a current frame coded according to a first mode. On a coder side, switching is at the junction between a previous frame coded according to a first coding mode and a current frame coded according to a second coding mode, a sound signal is filtered through a weighting filter to produce a weighted signal in the current frame, and a windowed zero-input response of the weighting filter is removed from the weighted signal. On a decoder side, a current frame of the target signal is first windowed, a left portion of a resulting window is skipped, and then a windowed zero-input response of the weighting filter is added to the decoded target signal to reconstruct the overlap-add target signal.

Abstract: A semiconductor apparatus includes a signal source 7 that outputs a signal of predetermined frequency, a frequency divider 15 that receives the output signal of the signal source and is capable of switching the output signal to two or more frequency division ratios, a delta-sigma modulator 16 that controls the frequency division ratio of the frequency divider, and a bandpass filter 17 that receives an output of the frequency divider. The frequency of the input signal of the frequency divider is divided by the frequency division ratio controlled by the delta-sigma modulator, and quantization noise appearing in the output of the frequency divider generated by the delta-sigma modulator is attenuated with the bandpass filter. The semiconductor apparatus easily can convert a signal output by a single signal source to a signal of predetermined frequency and supply a plurality of signals of predetermined frequency using a simple configuration with reduced chip size.

Abstract: A vehicle communication system has a receiving unit on a vehicle side for inputting a demodulated analog signal to determination units based on a received radio signal from a key unit. Then, one of the two determination units outputs a high-check signal that takes a high value when the analog signal is greater than a high threshold and the other determination unit outputs a low-check signal that takes a low value when the analog signal is smaller than a low threshold. Then, a sync. signal generator successively defines a determination period based on the high and low check signals. Then, a binary level of the analog signal is determined by a level determination unit based on the check signals, along with an estimation and a correction of an indefinite signal when the signal level cannot be determined based on a coding rule and the determination by the determination units.

Abstract: A system, apparatus and method for continuous synchronization of multiple ADC circuits is described. The ADC circuits can be arranged in a master-slave configuration within the system so that the converter clock is subdivided into slower speeds for the data output clock or for the control of de-multiplexing the outputs onto a wider bus, while maintaining ADC-to-ADC synchronization resilient to perturbations from noise and other upset sources. The configuration of the ADCs in the master-slave configuration can be varied according to overall system requirements in any one of a sequential configuration, a parallel configuration or a tree type of configuration, as well as others. Digital and/or analog timing adjustments can be made to each of the ADC circuits. The master clocking signals can be generated by a master clock generator circuit, which is either internally implemented in an ADC circuit, or externally implemented as a separate master clock generator circuit.

Abstract: A telecommunications transmitting device, including: an encoder configured to generate an initial signal formed by at least one reference sequence including an initial integer number of pulses dispatched over a same integer number of time windows; a transmitter configured to transmit the initial signal to a receiver; a demodulation device configured to determine that no acknowledgement from the receiver in response to the initial signal is received; and a processing unit configured to tune a second integer number of pulses constituting a second reference sequence, the second integer number being larger than the initial integer number, wherein the transmitter is configured to transmit a second signal formed by the second reference sequence including the second integer number of pulses dispatched over a same integer number of time windows.

Abstract: In a method and device for low-frequency emphasis, where the spectrum of a sound signal is transformed in a frequency domain and comprises transform coefficients grouped in a number of blocks, a maximum energy for one block having a position index is calculated. Also, a factor having a position index smaller than the position index of the block with maximum energy is calculated for each block. For each block, an energy of the block is calculated, the factor is computed from the calculated maximum energy and the computed energy of the block, and a gain is determined from the factor and applied to the transform coefficients of the block.

Abstract: An error model can be utilized to mitigate errors associated with a conversion system, such as an analog-to-digital or digital-to analog converter. The error model is adaptively calibrated to approximate error characteristics associated with at least a portion of the conversion system, such as a digital-to analog converter. The error model can be generated on-line during system operation or off-line to improve performance of various types of signal converters and systems using such signal converters.

Abstract: Circuit arrangement, LIN comprising such circuit arrangement as well as method for processing input signals of the LIN In order to further develop a circuit arrangement (100)—for processing at least one input signal (12) from at least one data bus (10) of at least one LIN and—for providing the data bus (10) with at least one output signal (18), as well as a corresponding operating method in such way that EMI performance and/or EMI performance of the LIN (300) is improved, it is proposed to provide—at least one analog-digital converting circuit (ADC) for converting the analog input signal (12) into at least one digital signal (14) to be processed, and—at least one digital-analog converting circuit (DAC) for converting the processed digital signal (16) into the analog output signal (18).

Abstract: A Huffman decoding method for decoding codewords included in an encoded data transmitted via a stream includes: processing codewords of a codebook that is predefined or extracted from the stream to derive at least an auxiliary lookup table, and storing each derived auxiliary lookup table in a storage device; and searching a decoded value corresponding to a target codeword included in the encoded data according to at least the auxiliary lookup table stored in the storage device and the codebook.

Abstract: An efficient method and system for detecting frame slip in an inband signalling block in pulse code modulation. The effect of frame slip on the inband signalling block is that the bits following the frame slip are transferred from the signalling block into an adjacent block. The octet slip is detected by searching an error bit in a signalling block by comparing it to a sample block. If an error bit is found, an error count for the adjacent block starting from the error bit is calculated. If the error count is more than one, a second error bit of the signalling block is searched (26) and bits of the adjacent block after second error bit are verified (27). If bits of the adjacent block after the second error bit are not correct, the octet slip cannot be assumed (29). Otherwise the octet slip can be assumed by analyzing error count and error bits.

Abstract: The present invention relates to a method for measuring radio channel quality in a radio communication system. In the method a modulated signal is received over a communication channel. The modulated signal has been modulated by using modulation parameters. A decoder decodes (305) the modulated signal and forms decoded data. The decoder creates (306) a decoder performance indicator (PS) that depends on the decoded data. Then a radio channel quality indicator (RCQI) is created, the radio channel quality indicator being essentially independent of the modulation parameters.

Abstract: A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.

Abstract: A mobile channel estimation algorithm is disclosed, for use in a DVB-H coded OFDM demodulator. The mobile channel estimation algorithm estimates the mobile channel transfer function, overcoming many of the delay spread limitations without using time filtering and without having to collect many OFDM symbols. Once the channel estimate has been found for each symbol, equalization and inter-carrier-interference cancellation may be carried out relatively easily using known methods. The mobile channel estimation algorithm may be used in DVB-H as well as in other areas of OFDM.

Abstract: A pulse pattern generator (11) has a pulse generating unit (12) and a lowpass filter (14), and the pulse generating unit (12) includes an amplitude-value setting unit (13, 13a). The pulse generating unit (12) generates a pulse signal formed in a step-like wave in which at least one of rise and fall of a signal is changed in a step-like manner. The lowpass filter (14) smoothes the pulse signal generated by the pulse generating unit (11), and outputs a smoothed pulse signal. The amplitude-value setting unit (13, 13a) adjusts an amplitude value of a step-like wave that forms the pulse signal in order to set a shape of an eye waveform based on a setting value when an eye pattern of the output from the lowpass filter (14) is formed. The pulse pattern generator (11) can output the pulse signal having a desired pulse pattern having the predetermined eye closure from the lowpass filter (14).

Abstract: The invention relates to techniques for implementing high-speed precoders, such as Tomlinson-Harashima (TH) precoders. In one aspect of the invention, look-ahead techniques are utilized to pipeline a TH precoder, resulting in a high-speed TH precoder. These techniques may be applied to pipeline various types of TH precoders, such as Finite Impulse Response (FIR) precoders and Infinite Impulse Response (IIR) precoders. In another aspect of the invention, parallel processing multiple non-pipelined TH precoders results in a high-speed parallel TH precoder design. Utilization of high-speed TH precoders may enable network providers to for example, operate 10 Gigabit Ethernet with copper cable rather than fiber optic cable.

Abstract: The present invention provides an approach for quadrature signal generation, which does not require orthogonal reference signals or nearly orthogonal reference signals as an input or given condition. The techniques provided herein can utilize a reference phase shift less than 90° but greater than 0°, along with an inversion to create orthogonal signals. The techniques provided here reduce the number of critical manipulations required from a hardware perspective.

Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.

Abstract: A system, apparatus and method for continuous synchronization of multiple ADC circuits is described. The ADC circuits can be arranged in a master-slave configuration within the system so that the converter clock is subdivided into slower speeds for the data output clock or for the control of de-multiplexing the outputs onto a wider bus, while maintaining ADC-to-ADC synchronization resilient to perturbations from noise and other upset sources. The configuration of the ADCs in the master-slave configuration can be varied according to overall system requirements in any one of a sequential configuration, a parallel configuration or a tree type of configuration, as well as others. Digital and/or analog timing adjustments can be made to each of the ADC circuits. The master clocking signals can be generated by a master clock generator circuit, which is either internally implemented in an ADC circuit, or externally implemented as a separate master clock generator circuit.

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 systems for transmitting a discrete information over a communication path comprising analogue communication lines and a pulse-code modulation (PCM) channel are considered, which methods and systems allow to increase the data transfer rate over the entire path up to the data transfer rate in the PCM channel irrespective of the quality of analogue communication lines really employed. In this case, the spectrum of signal being transmitted over the analogue communication line is selected such that the information transfer rate over this analogue communication line is equal to the information transfer rate over the PCM channel, and the number of quantization levels for samples being transmitted over the analogue communication line is the least possible for this analogue communication line.

Abstract: Methods and systems for processing a single sub-channel that includes two or more combined channels. Using intermediate frequency sub-sampling, two or more channels from a broad band signal are combined into a single sub-channel for further analog and digital processing. Each of the two or more channels is down converted to an intermediate frequency, filtered to remove certain undesired channels, and combined such that the two or more channels are adjacent to each other. A digital representation of the sub-channel is produced from the combined intermediate frequency channels. Each channel within the digital representation is down converted to baseband, and the in-phase and quadrature components are separated from each other. Compared to one direct down conversion technique, intermediate frequency sub-sampling as described in the application may reduce the number of analog to digital converters and control amplifiers used in analog processing by a factor of four.

Abstract: Bit-Edge Zero Forcing Equalizer. A novel solution is presented by which a BE-ZFE (Bit-Edge Zero Forcing Equalizer) is employed to drive an error term within a data signal to an essentially zero value. This new BE-ZFE looks at values of data that occur at the bit edges of a data signal and drives the associated error term to zero. The new BE-ZFE is appropriately implemented within communication systems that are phase (or jitter) noise limited. Some examples of such communication systems include high-speed serial links one type of which serviced using a SERDES (Serializer/De-serializer) where data that is originally in a parallel format is serialized into a serial data stream and then subsequently de-serialized back into a parallel data stream.

Abstract: A pulse modulator has a subtraction stage that produces a control error signal from the difference between a complex input signal and a feedback signal. A signal conversion stage converts the control error signal to a control signal. The control signal is multiplied by a complex mixing signal at the frequency ?0 in a first multiplication stage. At least one of the real and imaginary parts of the up-mixed control signal is then quantized by a quantization stage to produce a real pulsed signal. The pulsed signal is then employed to produce the feedback signal for the subtraction stage in a feedback unit. The pulse modulator according to the invention allows the range of reduced quantization noise to be shifted toward a desired operating frequency ?0.

Abstract: A codec circuit having a programmable digital bandpass filter is disclosed. According to one aspect, a bandpass filter matches the filter characteristics of the codec circuit to a transmitted pulse code modulation (PCM) signal. The codec circuit includes at least one programmable digital high-pass filter connected in series to at least one programmable digital low-pass filter. The setting filter characteristics for the programmable digital high-pass and low pass filters are each set by means of a signal identification device for identifying a PCM signal transmitted through the codec circuit. The filters are set as a function of the transmitted PCM signal in order to vary a bandpass filter characteristic for the programmable digital bandpass filter.

Abstract: A statistical value update unit calculates the fluctuation of measurement data. A filtering processing unit extracts a normal white noise component from the fluctuation of the measurement data using an adaptive lattice filter. A statistical test unit determines whether or not the variance of the normal white noise component is out of a predetermined scope in reference distribution. A change decision unit detects a stationary change of the status of a target system based on a detection ratio of an outlier.

Abstract: Any disagreement of the power level before encoding an audio signal and the power level after encoding the audio signal is adjusted to improve the sound quality to the auditory sense.

Abstract: A method of determining a variable quantization step size is disclosed. In the method of determining a variable quantization step size, a channel characteristic parameter is obtained in order to calculate a quantization step size (?) used in channel decoding. The quantization step size (?) is variably determined based on the channel characteristic parameter. Therefore, the method of determining a variable quantization step size may improve channel decoding.

Abstract: An apparatus for producing a fingerprint signal from an audio signal includes a means for calculating energy values for frequency bands of segments of the audio signal which are successive in time, so as to obtain, from the audio signal, a sequence of vectors of energy values, a means for scaling the energy values to obtain a sequence of scaled vectors, and a means for temporal filtering of the sequence of scaled vectors to obtain a filtered sequence which represents the fingerprint, or from which the fingerprint may be derived. Thus, a fingerprint is produced which is robust against disturbances due to problems associated with coding or with transmission channels, and which is especially suited for mobile radio applications.

Abstract: An analog/digital gain control device avoid some of the requirements associated with the nature of a closed-loop AGC circuits and which meets the remaining requirements without much difficulty uses an analog to digital conversion method that increases the number of effective ADC bits by compressing the baseband input analog signal using a logarithmic circuit. After the compressed analog signal is converted into a digital signal, a digital anti-log process or look-up table (LUT) is used to expand the digital signal back to the original linear scale. The word size of the output of the anti-log process is larger than the input word size due to the nature of the anti-log function. To reduce the word size of the digital signal an open loop normalization technique can be applied.

Abstract: An apparatus for compensating nonlinearly distorted multicarrier signals, a multicarrier signal receiver using the same, and a method therefor are provided, where the apparatus for compensating multicarrier signals and the multicarrier signal receiver using the apparatus extract parameter information on the HPA mode from the received signal so that nonlinear distortion of the received multicarrier signal is compensated for even though an accurate transfer function of a high power amplifier (HPA) is not known and side information or a special training signal is not transmitted when a signal is transmitted, such that nonlinearly distorted multicarrier signals such as OFDM signals transmitted by an HPA having a variety of transfer functions can be adaptively compensated and therefore a demodulated signal with an improved symbol error rate (SER) can be obtained.

Abstract: An analog/digital gain control device avoid some of the requirements associated with the nature of a closed-loop AGC circuits and which meets the remaining requirements without much difficulty uses an analog to digital conversion method that increases the number of effective ADC bits by compressing the baseband input analog signal using a logarithmic circuit. After the compressed analog signal is converted into a digital signal, a digital anti-log process or look-up table (LUT) is used to expand the digital signal back to the original linear scale. The word size of the output of the anti-log process is larger than the input word size due to the nature of the anti-log function. To reduce the word size of the digital signal an open loop normalization technique can be applied.

Abstract: Embodiments of the invention provide methods for determining the characteristics of a local loop using the multi-carrier tones of a DSL activation process. For one embodiment, one or more sets of downstream multi-carrier tones of a digital subscriber line activation process are received at an extended voice-band modem and are capable of transmitting upstream multi-carrier tones of the digital subscriber line activation process over a local loop. The downstream carrier tones and the upstream carrier tones are folded into a first multi-tone signal and a second multi-tone signal, respectively, each having a plurality of component signals within the spectrum band of the extended voice-based modem. The total signal power of the first multi-tone signal is divided by the power of one of the plurality of component signals of the second multi-tone signal to determine a ratio value. The ratio value is used to determine a characteristic of the local loop.

Abstract: An analog/digital gain control device avoid some of the requirements associated with the nature of a closed-loop AGC circuits and which meets the remaining requirements without much difficulty uses an analog to digital conversion method that increases the number of effective ADC bits by compressing the baseband input analog signal using a logarithmic circuit. After the compressed analog signal is converted into a digital signal, a digital anti-log process or look-up table (LUT) is used to expand the digital signal back to the original linear scale. The word size of the output of the anti-log process is larger than the input word size due to the nature of the anti-log function. To reduce the word size of the digital signal an open loop normalization technique can be applied.

Abstract: A digital wireless communications device enabling call distance to be increased while avoiding mute processing as far as possible, even if code error occurs in an ADPCM code. The digital wireless communications device may be configured to include: an error detector for detecting code error in an ADPCM code received via a wireless circuit; an ADPCM decoder for generating a PCM signal by decoding the ADPCM code; and a substitution unit for determining that a click noise is generated if the high-speed scale factor and the low-speed scale factor determined by the ADPCM decoder within a predetermined time period after the error detector has judged that there is a code error in the ADPCM code, and the received ADPCM code itself, respectively exceed prescribed threshold values, and for substituting the ADPCM code with a predetermined prescribed code.

Abstract: A method and apparatus is disclosed herein for decoding data (e.g., video data) using transforms. In one embodiment, the decoding process comprises scaling a block of coefficients using a scaling factor determined for each coefficient by computing an index for said each coefficient and indexing a look-up table (LUT) using the index. The index is based on a quantization parameter, a size of the block of coefficients, and a position of said each coefficient within the block. The method also comprises applying a transform to the block of scaled coefficients.