Abstract: Embodiments described herein may include example embodiments of a method, article and apparatus for compressing data utilizing combinatorial coding which may be used for communicating between two or more components connected to an interconnection medium (e.g., a bus) within a single computer or digital data processing system, and/or for communication between computing platforms via a network or other interconnection medium.

Abstract: Disclosed is a digital television system carrying out modulation/demodulation by VSB (vestigial side band) . A VSB transmitter includes an additional error correction encoder designed in a manner that a signal mapping of a TCM encoder is considered, a multiplexer (MUX), a TCM encoder operating in a manner corresponding to state transition processes of the additional error correction encoder, and a signal transmission part including an RF converter. And, A VSB receiver includes a signal receiver part receiving a signal transmitted from the transmitter, a TCM decoder, a signal processing part including a derandomizer, and an additional error correction decoder part.

Abstract: Disclosed is a digital television system carrying out modulation/demodulation by VSB(vestigial side band). A VSB transmitter includes an additional error correction encoder designed in a manner that a signal mapping of a TCM encoder is considered, a multiplexer(MUX), a TCM encoder operating in a manner corresponding to state transition processes of the additional error correction encoder, and a signal transmission part including an RF converter. And, A VSB receiver includes a signal receiver part receiving a signal transmitted from the transmitter, a TCM decoder, a signal processing part including a derandomizer, and an additional error correction decoder part.

Abstract: The system, apparatus and method of the present invention provides a single carrier block transmission with guard interval as a means of communications over multi-path channels that enables frequency domain equalization, and therefore, has many of the advantages of OFDM, but does not have some of the drawbacks such as high PAPR and the need for high resolution ADCs. While the use of guard intervals in single carrier communications enables low complexity detection of the signal, it reduces bandwidth efficiency. The present invention improves the bandwidth efficiency by adjusting the length of the guard interval adaptively. Also, by allowing both time-domain and frequency domain equalizations, besides improving bandwidth efficiency, the inventions allows for great flexibility in receiver design.

Abstract: In an analog signal compression method with digital compression and digital expansion of linear based or sine-based linear signals, a signal coordination point is determined in a bi-adaptive scaleable mV/step and a time/step structured plane. To accomplish this, the linear or sine-based input signal is digitized, the breaks of the digitized linear or sine-based signal are detected, and the time difference and the amplitude difference of two successive breaks of the linear or sine-based signal are determined. This permits time differences and amplitude differences of successive breaks to be value coded as a data word on the basis of adaptive scaleable time-per-step tables and voltage-per-step tables so that the time-per-step tables and the voltage-per-step tables are selected depending on the absolute value of the determined time difference and amplitude difference, thus producing compressed data. An associated expansion method for reconstructing the original analog signal is also disclosed.

Abstract: Embodiments of a method and apparatus for decoding signals are disclosed. The method includes receiving modulated signals, generating bits representing the signals, and associated reliability of each bit. The method further includes executing a first stage of decoding the bits using a first component code, and simultaneously executing the first stage of decoding again using a second component code, and executing a second stage of decoding using the first component code. The first and second stages of decoding are used to generate the bit stream. Another method includes receiving modulated signals, generating bits representing the signals, and associated reliability of each bit. The method further includes executing a first stage of N stages for decoding the bits, the first stage using a first of M component codes, and simultaneously executing a plurality of the N stages of decoding, each of the plurality of N stages using a different one of the M component codes.

Abstract: A source sends an encoded data block during a first time interval to a destination and to a relay. The source sends additional parity bits for the encoded data block during a second time interval. The relay partially decodes that encoded data block to a process-defined end point (typically only a partial decoding), such as a fixed number of decoding iterations. After partial decoding the relay forms a modified data block having corrected information bits and the parity bits of the block it received, and sends the modified data block to the destination, during the second time interval. The destination decodes to a results-defined end point the modified data block it received from the relay using the additional parity bits it received from the source. If that end-point cannot be reached, the destination may begin anew decoding the original encoded data block it received from the source.

Abstract: The present invention considers a direct sequence spread spectrum system wherein every user employs the same spreading code. In a preferred embodiment, received signal is correlated with the training signal. Peaks of correlator output are used to identify the delays corresponding to the significant multipath components. The delays that are within a predetermined number of chips of each other are associated with a hypothesized user. In an alternate embodiment, a user separation technique is used which is based on the observation that after the training period, different users send uncorrelated data. In another alternate embodiment, linear minimum mean squared error (LMMSE) based correlators locked to individual users are employed. These correlators only provide multipath components of their corresponding users as output.

Abstract: The invention relates to a method for the compression of data comprising values to be coded, in particular grey values or prediction errors using a run-length coding. A bit sequence of a bit plane which represents at least one item of partial information of at least one value to be coded is thereby coded coherently. This method is particularly used in connection with medical image data.

Abstract: A wireless device includes a case, user interface, an antenna, first processing resources, and second processing resources. The first processing resources couple to the antenna and to the second processing resources and receive and operate upon a data signal transmitted by a transmitting wireless device. In its processing operations, the first processing resources produces decoded data blocks and writes the decoded data blocks to a data buffer. When a processing resource threshold is met with regard to the plurality of decoded data blocks, the first processing resources issue a service processing interrupt to the second processing resources. In response to the service processing interrupt, the second processing resources retrieve the plurality of decoded data blocks from the data buffer in response to the service processing interrupt and process the plurality of decoded data blocks.

Abstract: A transcoder comprises a receiver (101) which receives input data representing an encoded signal and comprising first encoding data and first parametric extension data. The encoded data is fed to a decoder (103). The output of the decoder (103) is fed to an encoder (105) which generates second encoded data according to a different encoding protocol or with different encoding parameters. The first parametric extension data is fed to an extension data processor (109) which generates second parametric extension data directly from the first parametric extension data. The second encoded data and the second parametric extension data is combined in an output processor (107) to generate a transcoded signal comprising separately determined parametric extension data. The parametric extension data may be Spectral Band Replication (SBR) or Parametric Stereo (PS) extension data for an audio bitstream. Improved quality and reduced complexity is achieved by the separate transcoding of the parametric extension data.

Abstract: Hierarchical modulation preferably uses a 16-ary quadrature amplitude modulation (QAM) modulator with a convolutional encoder that combine to effectively provide unequal protection to two different segments of streaming input data, with improved power efficiency with the same bandwidth efficiency, by encoding the first segment as coded data as LSBs as the second segment remains uncoded as uncoded MSBs, with the MSBs used for QAM constellation modest reliability interquadrant demodulation and detection, and with the LSBs used for low reliability intraquadrant detection, but with the LSBs subject to convolutional encoding and decoding rendering the LSBs with high reliability detection, such that, the two segments have unequal coding and modulation for providing unequal levels of reliability detection.

Abstract: Techniques and tools are described for using quantization bias that accounts for relations between transform bins and quantization bins. The techniques and tools can be used to compensate for mismatch between transform bin boundaries and quantization bin boundaries during quantization. For example, in some embodiments, when a video encoder quantizes the DC coefficients of DC-only blocks, the encoder compensates for mismatches between transform bin boundaries and quantization bin boundaries. In some implementations, the mismatch compensation uses an offset table that accounts for the mismatches. In other embodiments, the encoder uses adjustable thresholds to control quantization bias.

Abstract: The invention concerns a method for interleaving a dedicated set of bits that is divided in an appropriate number of blocks using for each block an interleaver matrix with a predefined first dimension for interleaving the bits within each block, whereby said appropriate number of blocks is estimated in such a way that a value of a second dimension of the interleaver matrix that is needed to interleave the bits within each block lies within a dedicated range, a base station (BS1-BS8), a user terminal (T1-T4) and a communication network (CN) therefor.

Abstract: A single carrier transmission scheme which utilizes space-frequency block coding and frequency domain equalization (SF-SCFDE) is proposed for frequency selective and fast fading channel. It is shown that employing this technique in slow fading environment depicts the same performance as that obtained with space-time coding scheme. However, in the more difficult fast fading channels, the proposed scheme exhibits much better performance.

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: Computer readable medium, a system and a method for providing an image for use in a mobile device are provided. Object files are selected that include at least some code and some data for use by the mobile device. Components are generated from the object files. Whether a component is larger than a logical block is determined. The component is partitioned into modules in response to determining that the component is larger than the logical block. Unique addresses are established for components and any modules. A memory layout is generated including the unique addresses. Components and any modules are loaded at corresponding unique addresses to provide the image.

Abstract: A communication system (10) receives a communication signal comprising first and second data with different compression levels, such as highly compressed and weakly compressed levels. A mode detector (15) detects the level of compression. One or more signal decoders (20, 22) decode the highly compressed data. An analyzer (30) determines the type of enhancement required. One or more processors (48, 50, 80) enhance the data as required. An encoder (60) reencodes the enhanced decoded data. Metrics (90) may aid the operation of the analyzer (30). The communication system may include telephones (120, 122, 124, 126). Processors (103, 104) enhance signals in opposite first and second directions between pairs of the telephones. A path (106) connects the processors in tandem. One or more switches (101, 102) disable signal enhancement for one of the processors depending on the compression level of the signals to avoid degrading call quality.

Abstract: A DPCM operation which can reduce the size of differential data and a method and an apparatus for encoding data using the DPCM operation are provided. A method for generating differential data includes generating differential data by performing a DPCM operation on quantized data and generating predicted differential data by performing a predicted DPCM operation on the quantized data, generating circular-quantized differential data and circular-quantized predicted differential data by performing a circular quantization operation on the differential data and the predicted differential data so as to reduce their ranges, and selecting one of the circular-quantized differential data and the circular-quantized predicted differential data depending on their magnitudes.

Abstract: A transmitting terminal generates a stream of a given number of PCM codes whose polarity bits are each fixed at “1”, and transmits it to a receiving terminal. Subsequently, the transmitting terminal creates a data PCM stream made of full-bit PCM codes by fixing the polarity bit of each PCM code at “1” and by sequentially allocating the remaining bits to transmission data, and transmits it to the receiving terminal. The receiving terminal determines whether the polarity bit of each PCM code in a received PCM stream is “1”, and counts the number of consecutive PCM codes whose polarity bits are “1”. When the count value becomes equal to or more than the given number, reception data is rearranged from the remaining bits other than the polarity bit of each PCM code in a subsequently received PCM stream.

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: An apparatus for converting pulse code modulation (PCM) signals in a telecommunication system between two different modulation standards. In the apparatus, a channel selector generates a channel select signal for identifying at least one channel of the multiple channels containing input digital signals modulated by said one of said two different modulation standards. A first codec demodulates a signal on a channel designated by the channel select signal which is modulated by the first modulation standard, into an analog signal to be supplied to a second codec, which modulates the converted analog signal input into a digital signal using the second modulation standard. The reverse process is executed in a similar manner.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: This patent describes a method and system which overcomes the LO-leakage problem during modulation and demodulation, common to direct conversion and similar RF transmitters and receivers. This problem is solved using a virtual local oscillator (VLO™) signal which emulates mixing with a local oscillators (LO) signal. The VLO signal is constructed using complementary mixing signals that suppress mixing power in the bandwidth of the input signal, and within the bandwidth of the output frequency. Specifically, mixing is done in two or more stages, using time-varying mixing signals ?1 and ?2 which satisfy the following criteria: ?1*?2 having significant power at the frequency of the LO being emulated, one of ?1 and ?2 having minimal power around the frequency of the output signal y(t), and the other of ?1 and ?2 having minimal power around the center frequency, fRF, of the input signal x(t).

Abstract: A system is provided for that improves video quality and resolution. The system include a processor configured with logic to estimate an image from at least one observed image to produce an estimated image, the observed image having a lower spatial resolution than the estimated image. The processor is further configured with logic to simulate steps of an imaging process and compression process and update the estimated image with statistical and/or deterministic information if further optimization is needed, otherwise designate the estimated image as a solution.

Abstract: A communication system (10) receives a communication signal comprising first and second data with different compression levels, such as highly compressed and weakly compressed levels. A mode detector (15) detects the level of compression. One or more signal decoders (20, 22) decode the highly compressed data. An analyzer (30) determines the type of enhancement required. One or more processors (48, 50, 80) enhance the data as required. An encoder (60) reencodes the enhanced decoded data. Metrics (90) may aid the operation of the analyzer (30). The communication system may include telephones (120, 122, 124, 126). Processors (103, 104) enhance signals in opposite first and second directions between pairs of the telephones. A path (106) connects the processors in tandem. One or more switches (101, 102) disable signal enhancement for one of the processors depending on the compression level of the signals to avoid degrading call quality.

Abstract: Processes and apparatus for improving the state of the art for watermarking and data protection. The disclosure includes feature-based watermarks, auto- and cross-correlation techniques for determining scaling and rotation, transitions in time based watermarking, autocorrelation watermarks for images, and dynamic content scrambling of static files.

Abstract: Fractional bit rate encoding in a pulse amplitude modulation (PAM) communication environment allows the transmission of fractional bit rates, thus maximizing the use of signal-to-noise ratio (SNR) available on a communication channel. The invention allows the transmission of fractional bit rates in a PAM transceiver, thus allowing the encoding and transmission of a fractional number of bits on each PAM transmit symbol. By encoding a non-integer number of bits, a non power-of-two number of PAM levels can be encoded.

Abstract: A method of channel estimation in a Code Division Multiple Access (CDMA) transmission system that incorporates Pilot Symbol Assisted Modulation (PSAM) using an iterative coherent detection method to estimate the phase and frequency of the received pilot symbols. Arctangent calculations are used to estimate phase and frequency. An iterative least squares linearization identifies and corrects values of the arctangent associated with an incorrect 2? alias, which arise due to the multiple-valued nature of the arctangent function. An alternative non-iterative least squares linearization also corrects the arctangent values, based on a calculation involving stored values of the pilot symbols.

Abstract: A method and apparatus for Viterbi detector state metric re-normalization. The method includes fabricating a Viterbi detector (138) having a predetermined number of states, wherein the Viterbi detector (138) stores a state metric value and a branch metric value for each state, and wherein the Viterbi detector (138) implements a trellis diagram. The method includes constructing a Viterbi detector (138) which can support a state metric value having g+h′ number of bits. The number of bits needed to represent the branch metric value is represented by (g) and the additional number of bits needed to represent the state metric value is represented by (h′). The additional number of bits (h′) is less than the additional number of bits (h) determined using the following inequality: 2h−1−h≧K−1, wherein K represent the constraint length of the trellis diagram.

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: The present invention relates to a system and method which serves as a refinement in the criteria used to improve the performance of audio signal processing systems. More specifically, the present invention provides a method by which the frequency and magnitude of artifacts added to audio signal data in an encoder device can be reduced. The encoding device through which the audio signal passes includes a filter bank for filtering source audio data to produce frequency sub-bands, a psycho-acoustic modeler for calculating signal to masking ratios from the frequency sub-bands of the source audio data, and a bit allocator for assigning for using the signal to masking ratios to assign a finite number of bits to represent the frequency sub-bands. In the absence of a significant event, the bit allocator performs a pre-bit allocation procedure to prevent artifacts or discontinuities in the encoded audio data.

Abstract: A comparator (12) for compensating the pulse width of the drive control signal of a power amplifier (54) by using a sense signal of an offset voltage generated between the nodes (H, I) of the power amplifier (54) is provided to enlarge the time width from the OFF operation of a pair of switching elements (Q1, Q4) to the ON operation of a pair of switching elements (Q2, Q3), thereby preventing the inconvenience that the switching elements (Q1, Q2) simultaneously turn on and suppressing a through-current. The pulse width for turning on the switching elements (Q1, Q4) is so corrected as to be wider or narrower than the pulse width for turning on the switching elements (Q2, Q3), thereby adjusting the intensity of an impressing voltage and canceling the offset voltage.

Abstract: A method and apparatus of signal compression using adaptive pulse code modulation (ADPCM) with variable width frame is provided. The present invention enables a low distortion rate in the process of the ADPCM signal compression by installing a frame width modifier on the input of a code analyzer and corrector which repetitively receives compressed data and generates reference values to attain an optimal delta value for a given resolution condition that enables high compression ratio and data integrity in the data compression and decompression processes.

Abstract: The invention provides a method and system for adaptive point to multipoint wireless communication. The wireless physical layer and the wireless media-access-control (MAC) layer collectively include a set of parameters, which are adaptively modified by a base station controller for communication with a plurality of customer premises equipment. The base station controller adjusts communication with each customer premises equipment individually and adaptively in response to changes in characteristics of communication, including physical characteristics, amount of communication traffic, and nature of application for the communication traffic.

Abstract: A circuit that provides the root-mean-square (RMS) value of an input signal and that detects and independently recovers from an output fault condition is provided. The circuit includes reconfigurable circuitry that changes from normal operating mode to fault recovery mode when an output fault is detected. During fault recovery mode, the circuit provides a modified output signal that allows independent recovery from an output fault condition. Once recovery is complete, the circuit returns to normal operating mode and provides a DC output signal proportional to the RMS value of an AC input signal.

Abstract: A clock synchronization system for synchronizing a first communications device and a second communications device to enable digital communication there between. A first device generates a first clock signal Fa. A second device generates a second clock signal Fb2. The second device includes a first PLL circuit and a second PLL circuit. The first PLL circuit is adapted to increase Fa by a factor K to produce a signal Fak. The second PLL circuit is adapted to increase Fak by a factor L to produce a signal Fbn. The second PLL circuit is further adapted to decrease Fbn by a factor N to produce the signal Fb2. The first PLL circuit and the second PLL circuit are adapted to adjust the values of K, L, and N such that a frequency lock is achieved between Fa and Fb2. enabling digital communication between the first device and the second device without requiring a predetermined phase lock between Fa and Fb2.

Abstract: A communication system (10) receives a communication signal comprising first and second data with different compression levels, such as highly compressed and weakly compressed levels. A mode detector (15) detects the level of compression. One or more signal decoders (20, 22) decode the highly compressed data. An analyzer (30) determines the type of enhancement required. One or more processors (48, 50, 80) enhance the data as required. An encoder (60) reencodes the enhanced decoded data. Metrics (90) may aid the operation of the analyzer (30).

Abstract: A modem or other telephone device (102, 502) is configured to identify (or receive notification of) the companding law used by a destination modem or telephone device. The telephone device (102, 502) can then, using stored conversion tables, predict the inverse mapping from the converted encoding law. Having done so, the telephone device (102, 502) can identify distorted portions of the decoded signal and modify the original signal before it is sent to the encoding or encoding-converter unit.

Abstract: A data processing system for the compression and decompression of data using Differential Pulse Code Modulation, and optimized for fast execution using a parallel processing DSP such as the Texas Instruments TMS320C8X family. Decompression is speeded up over the methods known in the art by combining the VLC codes and the additional interval bits into one code, which is then used as an index into a Look Up Table that yields the final result in one step instead of the multiple operations required by current implementations.

Abstract: A hyperspectral image encoder (10) for compressing hyperspectral imagery includes a differential pulse code modulation (DPCM) loop (26) to perform data decorrelation in the spectral domain and a discrete wavelet transform (DWT) processing means (28) to perform decorrelation in the spatial domain. The DPCM loop (26) determines an error image between a present image at the input of the encoder (10) and a predicted image. The DWT processing means (28) then divides the error image into a plurality of frequency subbands and quantizes information within the subbands in accordance with a plurality of predetermined quantization states to provide a coded output signal. In one embodiment, an interband predictor (24) is provided to predict an image, for use in calculating the error image, using the coded output signal.

Abstract: A transmitting/receiving system and a receiving apparatus wherein a plurality of information signals having mutually different frequency bands processed through predetermined modulation are transmitted in time series from a transmitting apparatus via an electric power line. A desired information signal is selected out of the plural transmitted information signals, and the transmission quality of the selected information signal is measured. When the result representing the measured transmission quality is low, the signal being currently received is switched to the information signal having another frequency band indicating a better transmission quality.

Abstract: A double density channel bank architecture compatible with an existing D4 backplane and D4 channel units transmits or receive PCM data on the PAM bus, while simultaneously transmitting or receiving PCM data on the PCM bus. A double density channel unit (CU) provides data for two transmit and two receive T-1 timeslots to increase the capacity of a D4 channel bank from 48 channels to 96 channels. Channel bank timing of PCM data on the PAM bus provides isolation of PCM data and PAM signals on PAM bus. Double density transmit and receive units each support an additional DS1. The receive unit transmits a signal on both CU channels to indicate T-1 failure and to commence trunk processing.

Abstract: A method and apparatus are described for selecting a companding format of a series of pulse code modulated (PCM) data words encoded under either an A-law or &mgr;-law companding format. The method includes the steps of expanding the PCM words into an average waveform sample value using both the A-law and &mgr;-law companding formats and selecting the companding format providing the expanded average sample value with a smallest relative absolute value.

Abstract: A Viterbi decoding system interprets bits in received QAM constellations as many-valued parameters rather than binary valued parameters. It performs the Viterbi algorithm using these many-valued parameters to provide results superior to hard decision decoding. Rather than applying a hard 0-1 function to the QAM data, the system uses a non-stepped linear or curved transfer function to assign values to the bits. This results in performance superior to pure hard decision decoding and approaches that of soft decision decoding; moreover, it is applicable in many situations where soft decision decoding cannot be used.

Abstract: An encoding apparatus is provided for encoding a wideband digital signal, comprising an input (1) for receiving the wideband digital signal, a splitter unit (3) for splitting the wideband digital signal into a number of M sub signals and a formatting unit (7) for assembling digital information including said data reduced sub signals into a digital output signal having a format suitable for transmission or storage.

Abstract: A communication device is provided having a speech encoder and a speech decoder and being able to retrieve and store voice messages in a memory. The messages are stored in the memory according to a message format. This format is more compressed than the speech encoding format which is provided by the speech encoder. The device includes a frame interpolation block for decompressing a stored message and thereby creating a signal according to the speech encoding format. A frame decimation block is also provided for compressing a speech encoded signal thereby allowing a corresponding voice message to be stored in the memory according to the message format.

Abstract: A linecard codec (250) eliminates limitations of companded code disclosed which increases data rates over a public switched telephone network. The network links a plurality of subscribers and services providers through a central office facility (25) which includes at least one digital backplane (150). The codec (250) comprises an analog interface (152) to the network and a converter (258) coupled to the analog interface and configured to convert analog signals from subscribers to linear coded data (259). A RAM table (262) is used to map the linear coded data (259) to a predetermined coding scheme based on the values stored in the RAM table (262). The RAM table (262) stores mapping values that determine a mapping function between data transmitted by a service provider on the digital backplane (150) and data transmitted to a subscriber.

Abstract: In a tailbiting convolutional encoder the encoder (12) is initialized with a sequence of symbols corresponding to the final symbols of a sequence to be encoded, resulting in that the begin state and the final state of the encoder are the same. The problem of the convolutional code obtained by such a coder is that it requires a substantial computational resources to decode it. In the simplified decoder according to the invention, the decoding process is extended by a plurality of trellis sections to a length N+&ngr;. At the final trellis section, the state with the largest path metric is selected as the best state. This best state is used as staring point for a trace back operation to find an earlier state at N trellis section before the final trellis section. The earlier state found there is regarded as correct, having as consequence that the trellis should terminate also in said state. By performing a trace back from said state the source symbols can be found.

Abstract: To prevent cell-delay-variation of ATM cells to be transmitted and to compress numeral data without increasing CPU load, an ATM data controller for controlling data exchange between a data processing system and an ATM interface comprises a first FIFO (3) for buffering transmission data supplied from and received data to be supplied to the data processing system, a third FIFO (7) for buffering transmission ATM cells and received ATM cells, and a second FIFO (5) for buffering transmission cell data to be assembled into the transmission ATM cells and received cell data obtained by disassembling the received ATM cells. When the transmission data is numeral data, a data compressing/decompressing means (4) compresses the transmission data by deleting upper four bits of every byte of the transmission data, and an ATM interface control section (6) sets a data-compression flag in the cell-header.