Abstract: Presently known d=1 codes have long trains consisting of consecutive 2T runs and an overall high frequency of occurrence of the shortest 2T runs that reduce the performance of the bit detector By using a code with an MTR constraint of 2 an improvement in the bit detection is achieved. A code constructed in a systematic way that provides an MTR constraint of 2 is presented. A variation of such a code is disclosed where one sub-code is used, where coding states are divided into coding classes and where code words are divided into code word types. Then, for a given sub-code, an code word of type t can be concatenated with an code word of the next sub-code if said subsequent code word of said next sub-code belongs to one of coding states of the coding class with index Tmax+1 t.

Abstract: The present invention relates to SISO decoder for iteratively decoding a block of received information symbols (r), in particular for use in a turbo decoder, said block being divided into a number of windows of information symbols.

Abstract: This ID proposes synchronization patterns for RLL codes with a (repeated) minimum transition run (RMTR) constraint, where the synchronization pattern comprises a synchronization pattern-body that contains a characteristic bit-pattern that represents a violation of the RMTR constraint. Using a violation of the RMTR constraint allows for short synchronization patterns.

Abstract: For measuring the influence of noise on the talking quality of a telephone link in a telecommunications network, a talker speech signal (s(t)) and a degraded speech signal (s?(t)) are fed to an objective measurement device for obtaining an output signal (q) representing an estimated value of the talking quality. The degraded signal includes a returned signal (r(t)) originating from the network during transmission of the talker speech signal over the telephone link. The objective measurement provided by the device is a modified PSQM-like measurement, which is modified to include modelling of masking effects resulting from noise present in the returned signal. Preferably, the modelling includes noise suppression performed on a difference signal (D(t,f)) in a loudness density domain using noise estimation.

Abstract: A decoding method for decoding Low-Density Parity Check codes in transmission and recording systems includes a running minimum loop with the following iterative acts: reading a reliability value from the input sequence of input reliability values; comparing the reliability value with a stored value; and overwriting the stored value with the reliability value if the reliability value is smaller than the stored value.

Abstract: Receivers (1) for receiving encoded block signals and comprising processor systems (2) decode block signals by using Viterbi algorithms for finding a first candidate/path in a trellis (18) and by generating cost signals for finding further candidates/paths in said trellis (18). To reduce storage capacity, cost signals are combined for series of branches and cumulated cost is compared with thresholds. While searching for said further paths, a search for a further path is made in view of cumulated cost exceeding a threshold or not. The insight of indirectly constructing a list of candidates and the basic idea of more directly constructing said list brings more control. These receivers are less complex, and are further improved by introducing increasing thresholds per trellis and successive combining and different trellis directions for generating and cumulating cost signals and combining in a lexicographical order, and using a check sum for list reduction and by making error detections.

Abstract: A method of decoding possibly mutilated code words (r) of a code (C) includes decoding the differences (D) of a number (L?1) of pairs of possibly mutilated code words (rib, ri+1) to obtain estimates (u, v) for the differences of the corresponding pairs of code words (ci, ci+1), combining the estimates (u, v) to obtain a number (L) of at least two corrupted versions (wj) of a particular code word (c), forming a code vector (z) from the number (L) of corrupted versions (wj) of the particular code word (c) in each coordinate, decoding the code vector (z) to a decoded code word (c?) in the code (C), and using the generator matrix (G) to obtain the information word (m) and the address word (a) embedded in the decoded code word (c?).

Abstract: A method of determining the time relation between an original or input speech signal (10) and an output speech signal (15) affected by time warping in a communications system, such as a VoIP (Voice over Internet Protocol) system. Wherein corresponding speech bursts (11, 12; 16, 17) of the input (10) and output speech signal (15) are located in accordance with a predefined signal property thereof. The corresponding speech bursts (11, 12; 16, 17) thus located and time aligned (10, 30) for the correction of continuous and discontinuous warping effects. A performance estimate is generated by comparing the time aligned input and output speech signals (10, 30) applying cross-correlation techniques and PSQM (Perceptual Speech Quality Measure) or PSQM+ (Enhanced Perceptual Speech Quality Measure) techniques.

Abstract: A method of and a device for output based objective speech quality assessment, wherein a degraded output speech signal comprising a speech information portion, is compared (5) with a reference signal retrieved from the output speech signal. The reference signal is provided by perceptual approximation of the speech information portion of the output speech signal using a speech recoder (2) producing a reference speech signal of finite bitrate. In a preferred embodiment, the speech recorder (2) is a speech codec.

Abstract: According to an objective measurement technique for determining the quality of an output signal (X(t)) of an audio or voice signals-processing and/or -transporting system, the output signal and a reference signal (Y(t)) are mapped on representation signals (R(X), R(Y)) of a suitably chosen perception model. A differential signal (D(t)) of the representation signals is averaged, in a time-averaging device, in two steps by two averaging members connected in series, over time, whereby a quality signal Q is generated. The time averaging is carried out by the first and second averaging members at the local and the global levels, respectively, using various averaging functions, e.g., preferably Lebesgue paveraging functions having various p-values p1>p2. The invention advantageously provides a high correlation between a generated quality signal Q and the human quality perception, even in the event of brief disturbances.

Abstract: Objective measurement methods and devices for predicting perceptual quality of speech signals degraded in speech rocessing/transporting systems may have poor prediction results for degraded signals including extremely weak or silent portions. Improvement is achieved by applying a first scaling step in a pre-processing stage with a first scalins factor (S(Y+&Dgr;), which is a function of the reciprocal value of the power of the output signal increased by an adjustment value (&Dgr;), and by a second scaling step with a second scaling factor (S&agr;(Y+&Dgr;); S&agr;i(Y+&Dgr;i), with i=1, 2), which is substantially equal to the first scaling factor raised to an exponent having a adjustment value (&agr;) between zero and one. The second scaling step may be carried out on various locations in the device. The adjustment values are adjusted using test signals with well defined subjective quality scores.

Abstract: A method of and an arrangement for obtaining quality indicators for an objective assessment of a degraded or output video signal with respect to a reference or input video signal. The strength of edges or signal transitions in both the input and the output video signals are calculated to provide input and output edge signals. By processing the edge signals, introduced edges and omitted edges in the output edge signal are established. For each of the luminance and chrominance signals of a color video signal first and second quality indicators are obtained from normalized values of the introduced edges and omitted edges, relative to the output edge signal and the input edge signal normalized by first and second normalization factors, respectively. By using smeared Sobel filtering, correlation of calculated MOS and observed MOS by human test persons reaches a value of above 0.9.

Abstract: For measuring the influence of noise on the talking quality of a telephone link in a telecommunications network, a talker speech signal (s(t)) and a degraded speech signal (s′(t)) are fed to an objective measurement device (22) for obtaining an output signal (q) representing an estimated value of the talking quality. The degraded signal includes a returned signal (r(t)) originating from the network during transmission of the talker speech signal over the telephone link. The objective measurement carried out by the device is a modified PSQM-like measurement, which is modified as to include a modelling (32b) of masking effects in consequence of noise present in the returned signal. Preferably the modelling includes a noise suppression (42) carried out to a difference signal (D(t, f)) in the loudness density domain using a noise estimation (41).

Abstract: A method of and a system for data exchange over a data network (1), such as the Public Internet (PI), between a user party (7) and a destination party (13). The destination party (13) connects to the data network (1) and the user party (13) has access to the data network (1) through one of a plurality of intermediate data network service providers (4; 5; 6) connecting to the data network (1). In accordance with the invention a direct data link (16) may be established between the data network service providers (4; 5; 6) for alternatively routing the data exchange over this direct data link (16). The direct data link (16) is established through a switched data network (15), such as the Public Switched Telephone Network (PSTN), the Integrated Services Digital Network (ISDN), and an Asynchronous Transfer Mode network (ATM).

Abstract: A method of and a device for output based objective speech quality assessment, wherein a degraded output speech signal comprising a speech information portion, is compared (5) with a reference signal retrieved from the output speech signal. The reference signal is provided by perceptual approximation of the speech information portion of the output speech signal using a speech recoder (2) producing a reference speech signal of finite bitrate. In a preferred embodiment, the speech recorder (2) is a speech codec.

Abstract: According to an objective measurement technique for determining the quality of an output signal (X(t)) of an audio or voice signals-processing and/or -transporting system, the output signal and a reference signal (Y(t)) are mapped on representation signals (R(X), R(Y)) of a suitably chosen perception model. A differential signal (D(t)) of the representation signals is averaged, in a time-averaging device (17), in two steps by two averaging members (31, 32) connected in series, over time, whereby a quality signal Q is generated. The time averaging is carried out by the first and second averaging members at the local and the global levels, respectively, using various averaging functions, e.g., preferably Lebesgue p-averaging functions having various p-values p1>p2. Advantage: a high correlation between a generated quality signal Q and the human quality perception, even in the event of brief disturbances.

Abstract: The invention concerns a method and devices for the detection of errors, in particular transmission errors, in data streams and/or data packets. In order to better detect systematic errors in particular, the error detection function according to the invention is variable. The detection function is varied on the basis of the time and/or the data themselves, for example by assigning an individual variation value to each index (packet index), effectively varying the data themselves. The invention is particularly suitable for application to compressed data streams.

Abstract: An authentication system having a first device and a second device between which signals may be communicated. The first device includes a first memory for storing a definition signal and the first device generates a first signal to be communicated to the second device. The first device also generates a first alteration signal, and introduces an alteration into a first section of the definition signal stored in the first memory in accordance with the first alteration signal. The second device includes a second memory for storing the definition signal, and generates, in response to a receipt of the first signal, a second signal to be communicated to the first device. The second signal includes a second section of the definition signal stored in the second memory, and the second device also generates a second alteration signal and introduces an alteration into a third section of the definition signal stored in the second memory in accordance with the second alteration signal.

Abstract: Known methods for being capable of carrying out, with the same data carrier, various authentication processes are based on non-standardized data carriers (smart cards) on which there are stored two different access codes. Per first authentication process, a first user generates a first user code which is compared to a version, transformed by way of a transformation, of a first access code stored on the data carrier. Per second authentication process, a second user generates a second user code which is compared to a version, transformed by way of the transformation, of a second access code stored on the data carrier. By now having the transformation in this case take place, per other authentication process, by way of another transformation, there may be used, for different authentication processes, the same worldwide standardized data carrier on which there is stored only one access code.