Abstract: The present invention includes a novel audio recognition peripheral system and method. The audio recognition peripheral system comprises an audio recognition peripheral a programmable processor such as a microprocessor or microcontroller. In one embodiment, the audio recognition peripheral includes a feature extractor and vector processor. The feature extractor receives an audio signal and extracts recognition features. The extracted audio recognition features are transmitted to the programmable processor and processed in accordance with an audio recognition algorithm. During execution of the audio recognition algorithm, the programmable processor signals the audio recognition peripheral to perform vector operations. Thus, computationally intensive recognition operations are advantageously offloaded to the peripheral.

Abstract: A method for transcoding a CELP based compressed voice bitstream from source codec to destination codec. The method includes processing a source codec input CELP bitstream to unpack at least one or more CELP parameters from the input CELP bitstream and interpolating one or more of the plurality of unpacked CELP parameters from a source codec format to a destination codec format if a difference of one or more of a plurality of destination codec parameters including a frame size, a subframe size, and/or sampling rate of the destination codec format and one or more of a plurality of source codec parameters including a frame size, a subframe size, or sampling rate of the source codec format exist. The method includes encoding the one or more CELP parameters for the destination codec and processing a destination CELP bitstream by at least packing the one or more CELP parameters for the destination codec.

Abstract: The invention concerns audio signal processing, comprising: a first processing of an audio source signal, using at least a mathematical transform applied on first sequences of samples obtained by applying first segmentation windows on the audio source signal; and a second audio processing applied on second sequences of samples obtained by applying second segmentation windows on the signal delivered by the first step; the two successive first windows and/or the two successive second windows overlapping, the overlaps being such that the segmentations are synchronous.

Abstract: Compression of voice and data signals by at least an order of magnitude that permits greater use of the limited bandwidth at the low frequency of operation. A voice recognition system that converts words spoken into a microphone into a sequence of letters and gaps. An encoder codes the letters into a digital message. A transmitter transmits the digital message to a receiver over a communications link. The received digital message is decoded in a decoder and a speech synthesizer converts the decoded message into spoken words that are annunciated on a speaker. In addition, an initial message that identifies a stored voice type that is to be synthesized, or a simultaneous signal that tailors the voice synthesizer in real time as the voice of the speaker changes may be transmitted to cause the speech synthesizer to more accurately resemble a speaker's voice. A speech-to-text processor and a display may be used to display the message at the receiver for hearing impaired users and users located in noisy areas.

Abstract: A network processing system is described that is able to monitor IP network traffic, including the ability to perform trap and trace on IP communications flowing over the IP network. The network processing system is able to scan the entire contents of data packets passing through it, and to associate related data packets into discrete sessions, or flows, which allows the network processing system to search for predetermined search criteria contained within those flows. If a flow is found to contain a predetermined search criteria, the network processing system is able to maintain a record of the flow or to replicate the flow and save it or send it to another IP address for monitoring. The monitoring of a flow can include the entire contents of the flow, or any subset of information in the flow such as call identifying information.

Abstract: A vocoder (125) is initialized, prior to processing an initial batch of audio data, from parameters extracted from the first frame of audio data (308, 310, 320, 330, 332). In the instant embodiment, parameters affecting voice encoding, which are based on estimates of direct current bias, are used to program a high pass filter (253) incorporated in the vocoder (125).

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations, wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched.

Abstract: The data stream between the transcoders (TCE1, TCE2) of a mobile wireless system is subdivided into a first data stream with samples for transmission and a second data stream with signal parameters for reconstruction of user data and/or for signaling. Both data streams are transmitted at the same time in particular in a handshake phase. The invention permits an improvement in the quality of transmitted data, e.g., speech data in a GSM network in tandem operation between mobile subscribers, in particular during a handshake phase.

Abstract: A method and apparatus for noise suppression within a distributed speech recognition system is provided herein. Mel-frequency cepstral coefficients (MFCCs) values are converted to filter bank outputs (F′0 through F′22). The filter bank outputs are then used by a noise suppressor (303) for channel energy estimation, noise energy estimation, etc. Noise-suppression takes place on F′0 through F′22 and the noise-suppressed filter bank outputs F″0 through F″22 are converted back to MFCC values.

Abstract: There is provided an audio mode automatic switching method, which automatically recognizes kinds of input audios to automatically switch and output audio mode. The method includes the steps of: collecting sample audio data in advance, then analyzing a feature of the sample audio data and extracting features according to kinds of audios; and if a listening audio is inputted, pattern-matching a feature of the listening audio with the features according to the kinds of audios in the step (a) to determine the kind of the listening audio and automatically switch the audio mode according to the determined audio kind.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Abstract: A system and method for interfacing with a component located in a network environment is provided. A user in a network environment can connect to a device on the network and automatically learn at least one detail regarding the device software image details. Examples of the software image details may include software version number, size in bytes, device model/family name, software filename, interface hardware details, and supported software feature set such as Internet Protocol (IP), Internet Packet Exchange (IPX), and AppleTalk. The invention provides capability of determining whether the software image version or feature set is supported by a product which the user desires to use, suggesting an upgrade to an appropriate software version or feature set to accommodate the product if the current version is not supported by the product, and automatically upgrading the software if the user approves of such action.

Abstract: A first signal of two signals to be compared for similarity is divided into small areas and one small area is selected for calculating the correlation with a second signal using a correlative method. Then, the quantity of translation, expansion rate and similarity in an area where the similarity, which is the square of the correlation value, reaches its maximum, are found. Values based on the similarity are integrated at a position represented by the quantity of translation and expansion rate. Similar processing is performed with respect to all the small areas, and at a peak where the maximum integral value of the similarity is obtained, its magnitude is compared with a threshold value to evaluate the similarity. The small area voted for that peak can be extracted.

Abstract: A signal processor for effecting the conversion of an audio data signal from one format to another. The signal processor has a signal converter that can selectively acquire two operative modes, namely a first operative mode and a second operative mode. In the first operative mode, the signal converter transforms the audio data signal from one format to another and releases the converted audio data signal from the output of the signal processor. In the second operative mode, the signal converter is disabled and permits passage of the audio data signal to the output without conversion. The signal processor has a control unit for controlling the transition of the signal converter between operative modes. The control unit enables the signal converter to pass from the first operative mode to the second operative mode when at least one operating condition has been satisfied.

Abstract: A speech transcoder includes a codebook in which a plurality of algebraic codes conforming to a second encoding method to serve as conversion candidates of the algebraic code of a first speech code, and a limiting unit for limiting the plurality of algebraic codes stored in the algebraic codebook to at least one algebraic code having a value equal to that of embedded data embedded in a second speech code to limit the conversion candidates, a determination unit for determining an element code corresponding to a converted speech code from the limited conversion candidates.

Abstract: The excitation in a CELP-like speech coder is recursively calculated. For a given bitrate and a given complexity, the recursive approach described lowers the complexity with minimum impact on speech quality. The excitation signal is a sum of at least three vector terms, each vector term being a product of a codebook vector zk and an associated gain term gk. A first vector term g0z0 is determined that is representative of a target excitation vector x. Each remaining vector term is recursively determined as a vector term gkzk representative of the difference between the target excitation vector x and the sum of previously determined vector terms, ∑ i = 0 k - 1 ⁢ g i ⁢ z i .

Abstract: A method for determining a set of distortion measures in a pattern recognition process, where a sequence of feature vectors is formed from a digitized incoming signal to be recognized, said pattern recognition being based upon said set of distortion measures. The method comprises comparing (S10) a first feature vector in said sequence with a first number (M1) of templates from a set of templates representing candidate patterns, based on said comparison, selecting (S12) a second number (M2) of templates from said template set, the second number being smaller than the first number, and comparing (S14) a second feature vector only with said selected templates. The method can be implemented in a device for pattern recognition.

Abstract: Pattern recognition, wherein a sequence of feature vectors is formed from a digitized incoming signal, the feature vectors comprising feature vector components, and at least one feature vector is compared with templates of candidate patterns by computing a distortion measure.

Abstract: A base station assembles a frame including information bits at a vocoding rate for downlink transmission over a traffic channel as channel bits at a channel rate. The base station places at least one rate-indicating bit at a beginning of the frame for indicating the vocoding rate. The mobile station evaluates the downlink transmission with consideration of the vocoding rate indicated by the at least one rate-indicating bit. The mobile station can determine the vocoding rate by decoding the beginning of the frame to permit power control in less than one frame duration from initial receipt of the frame at the mobile station.

Abstract: A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: A CELP type voice encoding device and a CELP type encoding device. Both the CELP type encoding device and the CELP type encoding device have a noise code book that can be searched in two modes in accordance with linear predictive analysis results, a pitch gain and a pitch cycle, all of which are obtained as analysis results of an input voice. Also the number of pulses forming a noise code vector is switched between a first case where a variation in pitch cycle is small througtout continuous sub-frames and in a second case where the variation is not small througtout continuous sub-frames.

Abstract: A method of increasing satellite communication quality by using a MEO satellite constellation (12) and a LEO satellite constellation (14) in combination with a decision algorithm which selects the appropriate constellation to route a communication signal through. The decision algorithm can be embodied in three ways: gateway based (18), individual subscriber unit based (22) and satellite based (12, 14). The MEO constellation (12) and LEO (14) constellation may be cross-linked, allowing for switching of service between satellites, as needed, during a communication session.

Abstract: A method of insuring the accuracy of transmitted or stored digital data involves the use of a cyclical redundancy check (CRC) code. The method is particularly useful for ensuring the accuracy of frames transmitted between multi-mode vocoders. The method allows a different CRC code to be used for each mode of a transmitting multi-mode vocoder. A receiving multi-mode vocoder checks the CRC code against the CRC coding formulas of the various modes. If the CRC code is satisfied under any one of the modes, the frame is labeled as “good”. If the CRC code fails under all the modes, the frame is labeled as “bad”. If the bit frame includes bits for indicating the mode of the transmitting multi-mode vocoder, the receiving multi-mode vocoder checks the CRC code against the CRC coding formula for the indicated mode only. If the CRC code passes for the indicated mode, the frame is labeled as “good”, otherwise, the frame is labeled as “bad”.

Abstract: The system and method of the present invention comprises a compressed domain universal transcoder that transcodes a bit stream representing frames of data encoded according to a first compression standard (TDVC coding standard) to a bit stream representing frames of data according to a second compression standard (MELP coding standard). The method includes decoding a bit stream into a first set of parameters compatible with a first compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard without converting the first set of parameters to an analog or digital waveform representation. Lastly, the second set of parameters are encoded into a bit stream compatible with the second compression standard.

Abstract: A typical radio frame (300) comprises A, B, and C vocoded bits (304). At the end of each frame (300) A and B bits (305) are inserted from a previous frame. Thus, each frame not only comprises A, B, and C bits (304) for that frame, but also comprises those A and B bits (305) originally transmitted in a prior frame. Thus, each frame comprises high and low priority vocoded bits (304) from the current vocoder frame, and those higher priority bits from a preceding frame (305). By placing an inner CRC (302, 303) around the most important bits of the vocoded frame, even though a frame is erased (e.g. its outer CRC (301) failed) it can still be verified that the most important bits in the vocoded frame are correct. Since the class B and C bits can tolerate some errors, the vocoded frame can then play out if its inner CRC passes.

Abstract: In a method and apparatus for a differentiated voice output, systems existing in a vehicle, such as the on-board computer, the navigation system, and others, can be connected with a voice output device. The voice outputs of different systems can be differentiated by way of voice characteristics.

Abstract: A voice decoder detects channel errors and loss of cryptographic synchronization using the change in spectral energy between sequential frames. The change in energy between frames is determined between corresponding LSP's of said successive frames and summed together. A running average of the change in energy for a predetermined number of frames is maintained. Current voice frames are eliminated based on the difference between the change in energy associated with the current frame and the running average. Accordingly, offensive audio associated with such channel errors or cryptographic synchronization loss is eliminated.

Abstract: In a speech codec, the total number of transmitted bits is reduced to decrease the average amount of bit transmission by imparting a relatively large number of bits to the voiced speech having a crucial meaning in a speech interval and by sequentially decreasing the number of bits allocated to the unvoiced sound and to the background noise. To this end, such a system is provided which includes an rms calculating unit 2 for calculating a root means square value (effective value) of a filtered input speech signal supplied at an input terminal 1, a steady-state level calculating unit 3 for calculating the steady-state level of the effective value from the rms value, a divider 4 for dividing the output rms value of the rms calculating unit 2 by an output min_rms of the steady-state level calculating unit 3 to determine a quotient rmsg and a fuzzy inference unit 9 for outputting a decision flag decflag from a logarithmic amplitude difference wdif from a logarithmic amplitude difference calculating unit 8.

Abstract: The invention relates to a method and a device for changing the temporal length and/or the tone pitch of a discrete audio signal. For improving the sound quality in such a method, according to the invention it is proposed that the audio signal be split into at least two partial signals and, in each case, fed to a processing channel; that the temporal length and/or the tone pitch of the partial signals be changed separately in different ways; and that the separately-processed partial signals then be combined into an output signal. Alternatively, according to the invention it is proposed that the audio signal be fed to at least two parallel processing channel, that the temporal length and/or the tone pitch of the audio signals be changed separately in different ways, that the separately-processed audio signals be split into two partial signals in each case, and that an output signal then be formed through combination of, in each case, at least one partial signal of each processing channel.

Abstract: The present invention provides a method and apparatus for improving the audio quality of a signal by reducing the effect of mis-determining the frame rate of a frame. The method includes the steps of determining that the frame rate of the current frame of information is eighth rate (324/340), determining that the previous frame was a full rate frame (334) and resetting the filter states of a speech decoder (336). The method further comprises the steps of utilizing alternative symbol error thresholds based on the number of consecutive frames with the same frame rate (308/328).

Abstract: A method and system for selecting formant trajectories based on input speech and corresponding text data. The input speech is analyzed to obtain formant candidates for the respective time frame. The text data corresponding to the input speech is converted into a sequence of phonemes which are then time aligned such that each phoneme is temporally labeled with a corresponding segment of the input speech. Nominal formant frequencies are assigned to a center timing point of each phoneme and target formant trajectories are generated for each time frame by interpolating the nominal formant frequencies between adjacent phonemes. For each time frame, at least one formant candidate that is closest to the corresponding target formant trajectories is selected according to a minimum cost factor. The selected formant candidates are output for storage or further processing in subsequent speech applications.

Abstract: The processing volume for codebook search for vector quantization is diminished by sending data representing an envelope of spectral components of the harmonics from a spectrum evaluation unit 148 of a sinusoidal analytic encoder 114 to a vector quantizer 116 for vector quantization, so that the degree of similarity between an input vector and all code vectors stored in the codebook is found by approximation for pre-selecting a smaller number of code vectors. From these pre-selected code vectors, such a code vector minimizing an error with respect to the input vector is ultimately selected. In this manner, a smaller number of candidate code vectors are pre-selected by pre-selection involving simplified processing and subsequently subjected to ultimate selection with high precision.

Abstract: A lattice-structured multiple description vector quantization (LSMDVQ) encoder generates M descriptions of a signal to be encoded, each of the descriptions being transmittable over a corresponding one of M channels. The encoder is configured based at least in part on a distortion measure which is a function of a central distortion and at least one side distortion. For example, if M=2, the distortion measure may be an average mean-squared error (AMSE) function of the form ƒ(D0, D1, D2), where D0 is a central distortion resulting from reconstruction based on receipt of both a first and a second description, and D1 and D2 are side distortions resulting from reconstruction using only a first description and a second description, respectively. Further performance improvements may be obtained through perturbation of the lattice points.

Abstract: A vocoder (125) is initialized, prior to processing an initial batch of audio data, from parameters extracted from the first frame of audio data (308, 310, 320, 330, 332). In the instant embodiment, parameters affecting voice encoding, which are based on estimates of direct current bias, are used to program a high pass filter (253) incorporated in the vocoder (125).

Abstract: A method which comprises forming a first noise reduction frame (18) containing speech samples; which is windowed by a first window function. For the windowed frame, noise reduction is performed for producing a second noise reduction frame (19; 45). A speech coding frame (44) to be formed comprises noise-reduced samples of at least two successive second noise reduction frames (45, 46), partly summed with one another. On the basis of said speech coding frame (44), a set of speech coding parameters pj are determined. A lookahead part (42) of the speech coding frame is at least partly formed of a first slope (41), the first slope (10, 41) comprising a set of most recent noise-reduced samples of the second noise reduction frame, not summed with the samples of any other second noise reduction frame. The method reduces the delay caused by speech coding and noise reduction.

Abstract: A speech encoding/decoding apparatus. A speech encoding apparatus has a coding portion for receiving input information related to an uncoded signal representative of an original speech signal, the coding portion including a fixed coding portion for receiving the input information and producing a first coded signal estimate, and an adaptive coding portion for receiving the input information and producing a second coded signal estimate. A controller is connected to the fixed coding portion and the adaptive coding portion for receiving information indicative of speech characteristics of the uncoded signal and generates a control signal; and a code modifier receives the first coded signal estimate from the fixed coding portion and the control signal from the controller and produces a modified signal estimate.

Abstract: An apparatus is provided for checking the consistency between two training words which can be used in, for example, a speech recognition or verification system. Two training examples are aligned using a dynamic programming alignment process and an average frame score is calculated from the alignment results together with the worst score in a number of consecutive frames. These values are then compared with similar values obtained from training examples which are known to be consistent to determine if the training examples are consistent.

Abstract: A data stream between transcoders of a mobile wireless system is subdivided into a first data stream with samples for transmission and a second data stream with signal parameters for reconstruction of user data and/or for signaling. Both data streams are transmitted at the same time permitting an improvement in the quality of transmitted data, e.g. speech data in a GSM network in tandem operation between mobile subscribers.

Abstract: A CELP type voice encoding device is disclosed.

Abstract: Methods and apparatus for processing a transmitted voice signal include a centralized frame controller providing at least one boundary control signal to voice processing blocks and controlling the operation of the voice processing blocks on the transmitted voice signal based upon the boundary control signal.

Abstract: A zero delay continuous long term (LT) pre-processing method operable in a speech codec that introduces no delay. The present invention provides an elegant solution to perform long term (LT) pre-processing of the pitch lag of a speech signal to save a large number of bits required in various speech coding methods, including the code-excited linear prediction method. The present invention is ideal for speech coding standards and methods that any undesirable delay at the end of a speech frame of the speech signal. The present invention overcomes a significant limitation in the art of speech coding, in that, a speech coding system that performs the invention is operable while providing real time operation and introducing no delay whatsoever.

Abstract: A method for searching an excitation (or fixed) codebook in a speech coding system. In a speech coding system including a synthesis filter for synthesizing a speech signal, a fixed codebook searcher according to the present invention segments a speech signal frame into a plurality of subframes to generate an excitation signal to be used in a synthesis filter, segments again each of the subframes into a plurality of subgroups, and searches the respective subframes each comprised of a plurality of pulse position/amplitude combinations for pulses. The fixed codebook searcher searches the respective subgroups for a predetermine number of pulses having non-zero amplitude, and generates the searched pulses as an initial vector. Next, the fixed codebook searcher selects a pulse combination including at least one pulse among the pulses of the initial vector, and then substitutes pulses of the selected pulse combination for pulses in other positions in the subgroups.

Abstract: The present invention permits a combination of a scalable audio coder with the TNS technique. In a method for coding time signals sampled in a first sampling rate, second time signals are first generated whose sampling rate is smaller than the first sampling rate. The second time signals are then coded according to a first coding algorithm and written into a bit stream. The coded second time signals are, however, decoded again, and, like the first time signals, transformed into the frequency domain. From a spectral representation of the first time signals, TNS prediction coefficients are calculated. The transformed output signal of the coder/decoder with the first coding algorithm, like the spectral representation of the first time signal, undergoes a prediction over the frequency to obtain residual spectral values for both signals, though only the prediction coefficients calculated on the basis of the first time signals are used. These two signals are evaluated against each other.

Abstract: Formants, corresponding to input speech units based either on a known text or the results of a speech recognition procedure, are generated from a formant synthesizer. A frequency response is generated based on the synthesized formants. A second frequency response is generated based on a speech signal which is received and which corresponds to utterances of speech units. The synthesized formants are modified based on a comparison of the frequency response corresponding to the synthesized formants and specific proportional characteristics of a frequency response of the input speech signal. In one illustrative embodiment, the comparison is then recalculated and further modifications are made accordingly to improve accuracy. In one illustrative embodiment, time aligning and frequency warping are utilized as modification functions.

Abstract: A radio signal transceiver receiving at its input a speech signal and producing an output signal at a given output rate, the speech signal having undergone a source coding intended to sufficiently compress the input signal to obtain the desired output rate while an acceptable distortion ratio is maintained. In order to improve the compromise of transmission quality of the speech signal and transmission rate by selecting the optimum coder from the available coders, the transceiver comprises a measuring device for measuring the distortion of the output signal of a coder and a check circuit for comparing the estimated distortion with set values and deriving therefrom the optimum coder for the measured distortion.

Abstract: Drive sound source coding means, decoding means has a plurality of algebraic sound source coding means, decoding means having sound source position tables different in distribution lean of sound source position candidates in a frame, each algebraic sound source coding means, decoding means for referencing spectrum envelope information and coding the sound source of an input voice based on a sound source position selected from among the sound source position candidates in the sound source position table and a polarity and selection means for selecting the algebraic sound source coding means, decoding means with the smallest coding distortion from among the plurality of algebraic sound source coding means, decoding means and outputting code representing the drive sound source position output by the selected algebraic sound source coding means, and polarity.

Abstract: Encoding of prototype waveform components applicable to telecommunication systems provides improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates a codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions.

Abstract: It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify and provide a means for generating a set of parameters ideally suited for this operational mode selection. Third, it is an objective of the present invention to provide identification of two separate conditions that allow low rate coding with minimal sacrifice to quality. The two conditions are the coding of unvoiced speech and the coding of temporally masked speech. It is a fourth objective of the present invention to provide a method for dynamically adjusting the average output data rate of the speech coder with minimal impact on speech quality.

Abstract: This apparatus (1) is intended to be connected to a cellular telephone network which transmits data in frames (HTR). This apparatus comprises a data reconstructing device (30) triggered by a signal (BFR) which indicates received bad data of a frame before this frame is reconstructed. The data are reconstructed by means of established waveforms of correctly received preceding data. The frame is reconstructed by copying the estimated waveform (westi) as many times as necessary.