Abstract: A method of compressing sounds in mobile terminals according to the present invention transforms pulse code modulations (PCM) codes, which are source data of bell sounds using recorded sounds or voice memos and are generated by sampling the sounds, through applying a differential method and, then, compresses the PCM codes using Lempel Ziv Welch (LZW) compresses technique, thus reducing a storage space required for storing bell sounds using sounds or voice memos in mobile terminals. According to the present invention, compression efficiency is maximized upon using LZW algorithm by transforming PCM code through applying differential method, thereby increasing restoration efficiency of original sounds and heightening compression efficiency by about 50%, compared with the existing compression storage method using ADPCM.

Abstract: An electrical circuit to add and subtract differential input signals may be implemented using distributed differential transmission lines having a length substantially equal to one quarter of a wavelength of the differential input signals according to an effective electrical permeability of the transmission line. Alternatively, the electrical circuit may be implemented with lumped reactive elements.

Abstract: A novel apparatus and method of differential decoding for use in a communication system such as a cable system. The differential decoding mechanism of the present invention enables the use of the Chase algorithm for Reed Solomon (RS) codes (i.e. non-binary codes). The mechanism is well suited for use in systems employing QAM data modulation/demodulation techniques and that also incorporate use of a differential encoder such as in DOCSIS capable cable modem systems. The differential decoding mechanism is operative to analyze the input to the differential decoder and adjust the decoding action accordingly. The mechanism generates the first and second candidate constellation points needed by the Chase algorithm. Considering the differential encoding, there are four possible constellation candidates. The differential decoder reduces these four possible options to two by eliminating from consideration two of them.

Abstract: A device for performing predetermined processing on an input signal that may have a signal amplitude of more than one bit. The input signal is obtained by subjecting one-bit serial signals to predetermined signal processing, wherein the signal amplitude of more than one bit is converted to a one-bit serial signal by accumulating the signal amplitudes that exceed one bit, delaying the accumulated signal on the basis of the input signal, and outputting the accumulated signal.

Abstract: A technique for emulating differential signaling is disclosed. In one exemplary embodiment, the technique is realized by encoding a plurality of input signals so as to generate a plurality of encoded signals having a spatial run length of N, wherein N is an integer having a value of at least two. Each of the plurality of encoded signals is then transmitted over a transmission medium so as to provide a respective plurality of transmitted encoded signals. Each of the plurality of transmitted encoded signals is then compared with at least N neighboring others of the plurality of transmitted encoded signals so as to recover a representation of each of the plurality of encoded signals. Each of the plurality of recovered encoded signals is then decoded so as to generate a plurality of decoded signals representing the plurality of input signals.

Abstract: A distortion removal system for reducing the quantization distortion that results from quantization of a signal by sample-by-sample quantization techniques is disclosed. The distortion removal system uses the signal previously quantized with the sample-by-sample quantization techniques to determine an expected quantization distortion. The expected quantization distortion is determined for each of a plurality of frames of the signal. The expected quantization distortion is removed from each of the frames of the signal.

Abstract: Systems and methods are described for full bridge integral noise shaping for quantization of pulse width modulated signals. A method for full bridge integral noise shaping comprises: receiving a first and a second reference PWM signal; summing the first and second reference PWM signals with a quantization error correction; quantizing the sum into a first and a second output PWM signal; differentially integrating the first and second reference PWM signals and the first and second output PWM signals according to a full bridge integral noise shaping algorithm to obtain the quantization error correction. An apparatus for performing a full bridge integral noise shaping quantization of a pulse modulated signal, includes: a single-ended to differential conversion circuit; and a full bridge INS quantizer circuit, coupled to the single-ended to differential conversion circuit.

Abstract: A shaped-wave application section inputs a first pulse signal corresponding to transmission data output from a transmission data output section, and applies, to a modulation and amplification section, a signal wave corresponding to a second pulse signal such that the waveform of the first pulse signal is shaped. The pulse signal applied to the base of the transistor of the modulation and amplification section is amplitude-modulated into an amplitude-modulated wave in which the amplitude-modulated portion is strengthened, in accordance with the signal wave corresponding to the second pulse signal applied to the emitter thereof. The amplitude-modulated wave is supplied to an antenna section including a resonance circuit via the collector thereof, and is transmitted as an electromagnetic wave from the antenna section.

Abstract: A method and signal processing apparatus for reducing the number of bits of a digital input signal (Mi), includes adding a pseudo-random noise signal (Na) to the digital input signal (Mi) to obtain an intermediate signal (Di), the pseudo-random noise signal (Na) being defined by noise parameters (Np), and quantizing the intermediate signal (Di), having a word length of n bits, to a reduced word-length signal (Me) having a word length of m bits, n being larger than or equal to m. The method further includes quantizing the intermediate signal (Di) using a first transfer function which is non-linear, the first transfer function being defined by non-linear device parameters (NLDp).

Abstract: The number of pins on an integrated circuit chip is reduced by encoding control signals into a differential clock. The differential clock has two clock lines with complementary signals that together represent a clock. Control signals inside a clock-transmitting chip are input to an encoder which determines which control signal is being asserted or de-asserted. The encoder drives a clock-control signal that either forces both differential clock lines low or stops the differential clock from pulsing. A clock-receiving chip detects the both-low or stopped differential clock and determines which control signal was asserted or de-asserted. A phase-locked loop (PLL) in the receiver keeps an internal clock running even when the differential clock is missing pulses. A sequence of M1 missing clock pulses, followed by N1 clock pulses, followed by M2 missing pulses encodes the control signal, where M1, N1, and M2 are whole numbers.

Abstract: A video encoding/decoding system based on 3D wavelet decomposition and the human perceptual model is implemented. JND is applied in quantizer design to improve the subjective quality of compressed video. The 3D wavelet decomposition helps to remove spatial and temporal redundancy and provides scalability of video quality. In order to conceal the errors that may, occur under bad wireless channel conditions, a slicing method and a joint source channel coding scenario, that combines RCPC with CRC and utilizes the distortion information to allocate convolutional coding rates are proposed. A new subjective quality index based on JND is presented and used to evaluate the overall system performance at different signal to noise rations (SNR) and at different compression ratios. Due to the wide use of arithmetic coding (AC) in data compression, it is considered as a readily available unit in the video codec system for broadcasting.

Abstract: A technique for emulating differential signaling is disclosed. In one exemplary embodiment, the technique is realized by encoding a plurality of input signals so as to generate a plurality of encoded signals having a spatial run length of N, wherein N is an integer having a value of at least two. Each of the plurality of encoded signals is then transmitted over a transmission medium so as to provide a respective plurality of transmitted encoded signals. Each of the plurality of transmitted encoded signals is then compared with at least N neighboring others of the plurality of transmitted encoded signals so as to recover a representation of each of the plurality of encoded signals. Each of the plurality of recovered encoded signals is then decoded so as to generate a plurality of decoded signals representing the plurality of input signals.

Abstract: A method and apparatus that differentially encodes and decodes data symbols in dual domains is taught. Data packets are encoded, transmitted, and decoded during a plurality of symbol intervals on a plurality of sub-carriers. Encoding and decoding are accomplished across both the time and frequency domains such that the minimum number of carrier states are employed as reference only states that do not encode a symbol of data in and of themselves. A rule of adjacency is followed, both across time and frequency, so that decorrelation is minimized. Any modulation scheme that is applicable to differential encoding and decoding can be utilized. Communication systems that couple via radio waves, through metallic conductors, or over fiber optic paths can be employed.

Abstract: A first set of signals is transformed into a second set of signals having a more stable set of current requirements. The more stable current requirements of the second set of signals are achieved by encoding the second set of signals with either an equal number, nearly an equal number, a constant number, or nearly a constant number of logic ones and logic zeros. A communication channel is provided for carrying the second set of signals from the first node to a second node.

Abstract: Provided is a code correction circuit for presuming an error pattern of the received code when a received ADPCM code contains a click noise and correcting such error. A device for improving voice signal comprises a click noise detector (13) which judges that a received ADPCM code contains a click noise when a short interval average value of the absolute value of the received ADPCM code exceeds a predetermined value, and a code correction circuit (11) which presumes an error position of the bit of the received ADPCM code and corrects such error so as to minimize the variation in the short interval average value of the absolute value of the received ADPCM code when a transmission error is generated in the receiving ADPCM code judged to be containing a click noise.

Abstract: A method of encoding a digital data stream which comprises providing a first stream of clock pulses at a first rate; deriving a second stream of clock pulses from the first stream at a second rate which is an integer sub-multiple of the first rate and less than the minimum bit rate in the data stream; monitoring the data stream for changes in level therein; counting pulses in the first stream starting with the occurrence of a pulse in the second stream; capturing the pulse count when a change in level in the data stream is detected or when the next pulse in the second stream of clock pulses occurs; and upon the occurrence of this pulse in the second stream producing a time domain descriptor of the data by combining the count obtained with information as to the direction of the change in level in the data input stream.

Abstract: A filter system with reduced switch thermal noise includes an input circuit for receiving an input signal and a feedback signal and providing a signal representative of the difference; a filter circuit including at least an input sampling capacitor and switch which introduce thermal noise error; a feedback circuit responsive to the filter circuit for delivering to the input circuit the feedback signal; the input circuit also amplifying the difference signal before it is submitted to the filter circuit to reduce the input-referred thermal noise by a factor of approximately the gain of the amplification; and a ?? modulator using such a filter.

Abstract: A digital data modulator is coupled to a source of a digital data signal. An encoder encodes the digital data using a variable pulse width code. A pulse signal generator generates pulses representing edges of the encoded digital data signal. A carrier signal generator generates a carrier signal having carrier pulses corresponding to the pulses from the pulse signal generator. A corresponding digital data demodulator is coupled to a source of a modulated signal having carrier pulses spaced relative to each other to represent a variable pulse width encoded digital data signal. A detector generates a variable pulse width encoded signal in response to received carrier pulses. A decoder decodes the variable pulse width encoded signal to generate the digital data signal.

Abstract: An encoding/decoding apparatus and method for an orientation interpolator node are provided. The encoding/decoding apparatus for an orientation interpolator node, which provides rotational information of an object in a 3-dimensional space, is formed of “key information” which indicates a time when a rotational movement occurs and “key value information” which indicates a rotational posture corresponding to the time. A predictive coding apparatus is used in the encoding process for compressing the information of the key and key value information. Therefore, in encoding an input orientation interpolator node in an information compression method of a predictive encoding method, the efficiency of encoding is improved and an reverse rotation due to an encoding error from lossy coding can be corrected.

Abstract: A transcoder-codec circuit arrangement that supports voice-switched hands-free radio operation. A first register is arranged to store a value indicative of a peak signal in a receive signal path, a first attenuator is coupled to the receive path, a second register is arranged to store a value indicative of a peak signal in a transmit signal path, a second attenuator is coupled to the transmit signal path, and a digital signal processor is coupled to the first and second registers and configured and arranged to update the value in the first register with a present peak receive signal level if the value in the first register is less than the present peak receive signal level. The digital signal processor is further arranged to update the value in the second register to a present peak transmit signal level if the value in the second register is less than the present peak transmit signal level.

Abstract: A communication mode determination section 201 determines the communication mode based on the CIR measured by a CIR measurement section 219; a DRC signal creation section 202 creates a DRC signal with a number corresponding to the communication mode; and a DRC power controller 205 refers to a transmission power table 206 showing the correspondence between DRC numbers and transmission power, and, based on the transmission power of the pilot signal output from a pilot power controller 209, increases transmission power in proportion as the DRC signal indicates that downlink channel quality is good.

Abstract: A predictive decoder removes granular noise from the decoded signal by one of the following methods: discarding a predetermined number of least significant bits of the decoded signal; taking the sum of each two successive values of the decoded signal and dividing the sum by two to obtain an output signal; or resetting the decoded signal to a fixed value during inactive periods. Inactive periods are detected according to criteria involving an adaptively adjusted step size and a quantized difference signal, both of which are obtained in the decoding process.

Abstract: An adaptive sigma delta modulator has an input stage, a conventional sigma delta modulator, and adaptation stage, and an output stage. The input stage produces a difference signal representing the difference between an analog input signal and an adaptive signal, the amplitude of the analog input signal being in a first range [−a+a]. The conventional sigma delta modulator produces an intermediate digital output sequence representative of the difference signal, the amplitude of the intermediate digital output sequence being in a second range [−b+b], wherein b<a. The adaptation stage produces the adaptive feedback signal such that the amplitude of the adaptive signal keeps the difference signal within the second range [−b+b].

Abstract: A digital delta-sigma modulator for controlling a multi-modulus divider in a fractional-N frequency synthesizer features cascaded delta-sigma modulator stages in a feed-forward circuit topology to extend the possible multi-modulus control output values between the range of −1 to +1. A direct input receives an N-bit input control word which can be dithered, for example, by a sinewave in a two's complement format. The digital delta-sigma modulator can be of any type and includes cascaded accumulators and pipelined accumulator topologies.

Abstract: The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.

Abstract: A power efficient and reduced electromagnetic interference (EMI) emissions transmitter for unshielded twisted pair (UTP) data communication applications. Transmit data is processed by a digital filter. The digital filter output data is converted to a current-mode analog waveform by a digital-to-analog converter (DAC). The digital filter is integrated with the DAC binary decoder in a memory device such as a ROM with time multiplexed output. DAC line driver cells are adaptively configurable to operate in either a class-A or a class-B mode depending on the desired operational modality. A discrete-time analog filter is integrated with the DAC line driver to provide additional EMI emissions suppression. An adaptive electronic transmission signal cancellation circuit separates transmit data from receive data in a bidirectional communication system operating in full duplex mode.

Abstract: A simplified method for frequency offset estimation in a TDMA cellular PCS environment using &pgr;/4-shifted DQPSK comprises the steps of multiplying a complex conjugate of a received complex-valued symbol and a succeeding symbol to produce a comparison vector having an angle equal to the phase angle between the received complex-valued symbol and the succeeding symbol, rotating the comparison vector so that the angle thereof is between 0° and 90°, and estimating the frequency offset by determining a constant deviation of the phase angle from an ideal phase angle value of 45° by calculating an average phase angle for a plurality of successive comparison vectors or correlating the rotated comparision vector against a bank of unit vectors to determine a maximum correlation.

Abstract: A method and apparatus for reducing the audible “clicks” or “pops” which occur when an ADPCM encoding and decoding system is employed in a communications system in which communication occurs over a dispersive channel. A novel technique is employed in which ADPCM-encoded silence is substituted for error-containing frames, and post-processing is performed on decoded frames while a muting window is open.

Abstract: Voice recording and playback mode using the G.726 half-rate within the personal handy phone system (PHS). When a portable station within the PHS operates as a voice recorder (e.g., functioning as an answering machine), a cost effective system in accordance with the present invention is adapted to compress and store received voice/sound signals in order to increase the usage of limited memory resources provided within the portable station. The present invention also enables previously compressed and stored voice/sound signals to be decompressed and played back in various portable station playback modes. Specifically, the portable station receives a voice/sound signal in a full rate (e.g., 32 kilobits-per-second) 4-bit adaptive differential pulse code modulation (ADPCM) data format in compliance with the International Telecommunication Union (ITU) recommendation G.726.

Abstract: A predictor for differential pulse code modulation that designates a subset averaged median (SAM) predictor, which removes transmission errors and/or minimizes error propagation. The SAM predictor in a preferred embodiment operates according to SAM ⁢   ⁢ ( X ) = ∑ i = 1 P ⁢   ⁢ a i ⁢ F i ⁢ ( X ) where X is an input vector within a predictor window, P is the number of median subfilters, ai is an optionally selected coefficient and Fi( ) is a feature equation of an ith median subfilter.

Abstract: A method of determining an integral portion of a carrier offset &Dgr;fc of an RF signal transmitted from a transmitter at a transmit carrier frequency fct and an apparatus for carrying out the method. The signal consists of at least two data symbols S1 and S2, each having a useful part preceded by a cyclic prefix containing a tail portion of the useful part, such that in the time domain the useful part occupies a symbol interval Ts and the cyclic prefix occupies a guard interval Tg. The carrier offset &Dgr;fc between a receive carrier frequency fcr and the transmit carrier frequency fct is calculated in the form of an integral multiple of the inverse 1/Ts of the symbol interval. The method is especially useful in application to data symbols which are multiplexed by the orthogonal frequency division multiplexing (OFDM) and are constructed from sub-symbols ck belonging to a 2m-ary constellation of complex values equally spaced in phase, such as phase-shift keyed (PSK) constellations, e.g.

Abstract: An adaptive sigma delta modulator has an input stage, a conventional sigma delta modulator, and adaptation stage, and an output stage. The input stage produces a difference signal representing the difference between an analog input signal and an adaptive signal, the amplitude of the analog input signal being in a first range [−a +a]. The conventional sigma delta modulator produces an intermediate digital output sequence representative of the difference signal, the amplitude of the intermediate digital output sequence being in a second range [−b +b], wherein b<a. The adaptation stage produces the adaptive feedback signal such that the amplitude of the adaptive signal keeps the difference signal within the second range [−b +b].

Abstract: A device for seamlessly providing 10BASE-T compatible data communications over an ordinary single twisted pair home phone line between multiple computers, between computers and peripherals, and between multiple peripherals. Each component that is to communicate over the home phone line will have a 10BASE-T compatible network interface card (NIC) for interfacing with the device. A transmit/receive switch is used to switch the device between a transmit mode and a receive mode. When signal are being transmitted from a component a Manchester coder decodes signals received from the NIC. A differential converter is used to convert the differential signal received from the NIC to a single signal. A modulator is used to modulate the signal to a RF signal using a modulation scheme such as PSK, QPSK, QAM or MCM schemes. A filter is used to limit the bandwidth of the modulated signal and a driver is used to amplify the signal to match the impedance of the phone line.

Abstract: A power efficient and reduced electromagnetic interference (EMI) emissions transmitter for unshielded twisted pair (UTP) data communication applications. Transmit data is processed by a digital filter. The digital filter output data is converted to a current-mode analog waveform by a digital-to-analog converter (DAC). The digital filter is integrated with the DAC binary decoder in a memory device such as a ROM with time multiplexed output. DAC line driver cells are adaptively configurable to operate in either a class-A or a class-B mode depending on the desired operational modality. A discrete-time analog filter is integrated with the DAC line driver to provide additional EMI emissions suppression. An adaptive electronic transmission signal cancellation circuit separates transmit data from receive data in a bidirectional communication system operating in full duplex mode.

Abstract: The system preferably includes a unique transmitter that sends both clock and data signals over the same transmission line. The receiver uses the same transmission line to send data signals back to the transmitter. The transmitter comprises a clock generator, a decoder and a line interface. The clock generator produces a clock signal that includes a variable position falling edge. The falling edge position is decoded by the receiver to extract data from the clock signal. The receiver comprises a clock re-generator, a data decoder and a return channel encoder. The clock re-generator monitors the transmission line, receives signals, filters them and generates a clock signal at the receiver from the signal on the transmission line. The return channel encoder generates signals and asserts them on the transmission line. The signal is asserted or superimposed over the clock & data signal provided by the transmitter.

Abstract: A digital transmission system (1) comprises: a transmitter (2), a receiver (3), and a transmission channel (4) coupled there between. Both the transmitter (2) and the receiver (3) are provided with an encoder (5) and a decoder (6) respectively, wherein a multilevel input signal is coded such, that a DC-balanced digital channel code is transmitted. The coders (5, 6) are embodied to match levels of the DC-free multilevel input signal and code words of the DC-balanced digital channel code such, that disparities of the selected code words are symmetrically grouped around zero disparity. This results in a DC-free digital channel code signal having a limited transmission band width, which signal can be conveyed with minimal hardware over the usually AC-coupled transmission channels.

Abstract: A method for phase encoding DPSK and N-PSK which maintains phase continuity and which does not require any reference symbols when using N-PSK. A sequence of known DPSK phases are encoded with respect to an initial phase. The N-PSK phases are then all offset by this same initial phase. The known DPSK phases can be used at a receiver to determine a sum consisting of the initial phase and any further phase shift introduced by the channel, and this sum is used to decode the unknown N-PSK phases. The method is more generally applicable to any case where a switch between a phase encoding method which requires an absolute phase reference and a phase encoding method which does not have an absolute phase reference is to be implemented on a single carrier or channel.

Abstract: In a PCM modem system in which equivalence classes are used to communicate information from a transmitter to a receiver, a method is provided to solve the problem of 180° phase reversals in the communications channel which result in a garbled transmission. This is accomplished by remapping the equivalence classes into a form that can be differentially encoded and decoded such that equivalence class identity is not lost during a phase reversal of the channel.

Abstract: An interface module includes a high density analog interface (HAI) for electrical interconnection between a programmable logic controller (PLC) and an analog to digital converter (ADC) or a digital to analog converter (DAC). The HAI includes a single application specific integrated circuit having a data scaling function block, a diagnostics function block configured to verify functionality of said scaling function block, a self-calibration function block configured to compensate for drift in said ADC, and a shared interface function block configured to electronically connect said module with a programmable logic controller (PLC).

Abstract: Method of adaptive differential pulse code modulation, ADPCM, which fundamentally satisfies the recommendations G.726 or G.727 of the CCITT, being implemented in an ADPCM encoder whereby, given an input sample, a quantified difference signal (DQ) is obtained by subtracting from the input sample an estimated value of this sample, the estimated signal (SE), whereby a feedback loop is formed which includes a prediction module (27). The prediction module (27) calculates, in the preceding frame, the estimated signal (SE) and a partial estimated signal of the sixth order prediction for the current frame sample. Consequently these operations are carried out in parallel with other of the algorithm, resulting in an overall speeding up of the encoding process.

Abstract: Digital data transmission using three logic states on a differential pair of signal lines. The three states are: a first line a threshold higher than a second line, the second line a threshold higher than the first line, and when both lines are approximately equal. The presence of three states allows the receiving circuit to recognize the beginning and end of a valid data bit. A receiving circuit using two comparators to generate strobes for latching the data is also disclosed. The strobes also clock a counter whose output is fed to a decoder. The output of the decoder is used to select one of N latches that are used to latch the incoming data.

Abstract: A method of coding information for transmission over a communication channel involves differentially coding selected bits of an input sequence with respect to bits of a previous input symbol to generate a transmit sequence comprising a plurality of transmit symbols. The differential coding method can be used in combination with unequal error protection and interleaving to protect bits during transmission.

Abstract: Voice recording and playback mode using the G.726 half-rate within the personal handy phone system (PHS). When a portable station within the PHS operates as a voice recorder (e.g., functioning as an answering machine), a cost effective system in accordance with the present invention is adapted to compress and store received voice/sound signals in order to increase the usage of limited memory resources provided within the portable station. The present invention also enables previously compressed and stored voice/sound signals to be decompressed and played back in various portable station playback modes. Specifically, the portable station receives a voice/sound signal in a full rate (e.g., 32 kilobits-per-second) 4-bit adaptive differential pulse code modulation (ADPCM) data format in compliance with the International Telecommunication Union (ITU) recommendation G.726.

Abstract: A device having alternative intercommunication formats, and an automated method of selecting the appropriate format to use, in dependence upon the ability of the device to which it is interconnected to utilize this format. Many devices utilize standard analog formats, such as NTSC, PAL, SECAM, and others, to communicate with other devices; many of these devices utilize digital formats, such as MPEG and others, for internal processing or storage. By appending a supplemental signal to the analog signal, the ability of the device to use a digital format can be communicated to the device to which it is interconnected. By communicating their capabilities, devices which have the ability to utilize the same digital format can automatically switch to this digital format for intercommunication, thereby avoiding the signal degradations typically associated with conversions to and from an analog form.

Abstract: This invention relates to duplex and to half duplex transmission schemes and more particularly to the application of such schemes in TDMA and other multiple access environments. In accordance with a first aspect of the invention, there is provided a method of multiplexing signals transmitted by a number of terminals onto a communications link; the link having a capacity less than the total capacity of the terminals; the method including the step of varying the rate of encoding of said signals transmitted by said terminals whereby the bandwidth o the transmitted signals corresponds to the portions allocated to the terminals. In accordance with another aspect of the invention, there is provided a system for the communication of signals in real time over a channel; said system adapted to vary the bandwidth of at least one signal such that the number of signals multiplexed onto said channel can be dynamically varied.

Abstract: A digital cordless telecommunications unit that serves for communications when paired with a similar unit and connected with a network is disclosed. The unit receives analog receive voice signals and transmits analog transmit voice signals. In addition, the unit transmits digital baseband transmit signals and receives digital formatted baseband receive signals. The unit includes a baseband chip, as well as an audio functions circuit and a system control functions circuit. The audio functions circuit comprises an audio front end for receiving the analog receive voice signals and transmitting the analog transmit voice signals, and an adaptive differential pulse code modulator codec, connected to the audio front end, for converting the analog receive voice signals to the digital transmit signals and converting the digital formatted baseband receive signals to the analog transmit voice signals.

Abstract: The quality of service provided by a connection in transferring data units (19) between first and second points (A, B) across a network (10), is measured by a method involving monitoring the normal connection traffic at the first and second points (A, B) to detect the occurrence of the same events at each point (unit 22). An event is deemed to have occurred at a monitored network point when a predetermined set of criteria concerning one or more data units is satisfied by the connection data-unit traffic at the point concerned. Whenever an event is detected, an event report or "digest" is generated (unit 25), this digest including a signature based on the contents of the data units giving rise to the event. Digests from both monitored network points (A, B) are sent to a correlation unit (30) where digests relating to the occurrence of the same event at the two network points are matched up. The matched digest pairs are then passed to a measurement unit (31) to derive quality of service measurements.

Abstract: A data communications system includes a standard telecommunications apparatus (1) arranged to transport analogue signals in a form of Adaptive Differential Pulse Code Modulated (ADPCM) digital signals generated from Pulse Code Modulated (PCM) digital signals representative of the analogue signals. A mapping data processor (13) coupled to a source (14) of digital data and to an ADPCM encoder (3) of the standard telecommunications apparatus (1), generates PCM coded symbols representative of the digital data in accordance with a predetermined mapping function. An unmapping data processor (15) connected to an ADPCM to PCM decoder (10) of the apparatus (1) operates to re-generate the digital data from communicated PCM signals.

Abstract: The present invention relates to arrangements and methods for radio frequency modulation and power amplification of an information signal (X.sub.IF). Modulation and amplification of the information signal (X.sub.IF) is carried out with very high linearity and efficiency over a wide frequency band by sigma-delta modulating (410) said information signal (X.sub.IF), whereby a digital signal (Y) is generated. The digital signal (Y) is up-mixed and voltage switched (420) by voltage-fed switches (423) controlled by the up-mixed digital signal (RF). Said switches are preferably conductive elements whose electric conductivity depend on the amount of ionising radiation that falls on the elements.

Abstract: A method and apparatus for performing adaptive differential pulse code modulation. The method and apparatus is particularly adapted for compliance with the ITU-T G.726 international standard. Certain intermediate values, including Y and SE, needed to compress or decompress a given sample are pre-calculated prior to receipt of the actual sample to which they correspond. Accordingly, when the sample is received, intermediate variables Y and SE are essentially immediately available. This allows the output value SR (during a decompression cycle) or I (during a compression cycle) to be available one clock cycle after receipt of a sample. The remaining clock cycles corresponding to that sample period are used to precalculate the intermediate variables Y and SE for the next sample. The method and apparatus requires only fifteen clock cycles per conversion.