WO2003096538A2 - Procede et dispositif pour recuperer un mot de code a partir d'un mot de code errone reçu et dispositif pour produire un mot de code, et systeme de controle - Google Patents

Procede et dispositif pour recuperer un mot de code a partir d'un mot de code errone reçu et dispositif pour produire un mot de code, et systeme de controle Download PDF

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Publication number
WO2003096538A2
WO2003096538A2 PCT/EP2003/004758 EP0304758W WO03096538A2 WO 2003096538 A2 WO2003096538 A2 WO 2003096538A2 EP 0304758 W EP0304758 W EP 0304758W WO 03096538 A2 WO03096538 A2 WO 03096538A2
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Prior art keywords
code
code word
packet
units
unit
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PCT/EP2003/004758
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German (de)
English (en)
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WO2003096538A3 (fr
Inventor
Markus Kaindl
Joachim Hagenauer
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Technische Universitaet Muenchen
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Technische Universitaet Muenchen
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Priority to AU2003226827A priority Critical patent/AU2003226827A1/en
Publication of WO2003096538A2 publication Critical patent/WO2003096538A2/fr
Publication of WO2003096538A3 publication Critical patent/WO2003096538A3/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
    • H04L1/0056—Systems characterized by the type of code used
    • H04L1/0067—Rate matching
    • H04L1/0068—Rate matching by puncturing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
    • H04L1/0045—Arrangements at the receiver end
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/004—Arrangements for detecting or preventing errors in the information received by using forward error control
    • H04L1/0056—Systems characterized by the type of code used
    • H04L1/0059—Convolutional codes

Definitions

  • the present invention relates to a method and apparatus for recovering code words from a received defective code word, and more particularly to a method and apparatus for recovering a code word transmitted over a lossy packet-oriented connection network and comprising a plurality of packets , Furthermore, the present invention relates to a method and to an apparatus for generating a plurality of packets for transmission over a packet-oriented connection network and to a transmission system.
  • the present invention is particularly suitable for the transmission of multimedia data via a lossy channel, channels in particular being considered in which both packet losses and individual, randomly distributed bit errors occur in the data stream.
  • the packet-based transmission of multimedia data via a communication system which consists of a line-bound core network (fixed network) and a radio network (mobile network), is considered below.
  • the core network bridges long distances, while the radio network ensures that a mobile subscriber can be reached wirelessly on the "last mile”.
  • a concrete example of such a transmission system would be e.g. B. the combination of the Internet with a modern mobile radio system, e.g. B. GSM-GPRS / UMTS, which is often referred to as mobile Internet.
  • a modern mobile radio system e.g. B. GSM-GPRS / UMTS, which is often referred to as mobile Internet.
  • the receiver is thus confronted with a data stream in which packets may be missing on the one hand and packets received on the other hand may contain individual bit errors.
  • Methods of this class aim to recognize the transmission errors on the receiver side and to conceal them by means of a suitable signal interpolation. Although these methods deliver very good results with low bit error rates or individual packet losses, they usually fail with higher bit error rates and with successive packet losses.
  • Interleaving / Multiplexing at the Sender Methods of this class spread the data of a packet over several packets (interleaving) or bundle the data streams of several participants (multiplexing) before transmission. This is associated with a delayed transmission of the data
  • Channel coding Methods of this class specifically insert redundancy (channel coding) into the data stream at the transmitter, which can be used to detect and correct transmission errors at the receiver.
  • Methods of this class generate several, usually two, descriptions of the same from a data packet. If only a part of the descriptions is now received, decoding with reduced quality can take place. If all descriptions are received, the decoding takes place in the original quality. These methods are particularly suitable for transmission via packet loss channels, but have so far found little use.
  • the present invention is based on the object of making it possible to transmit data with improved quality in a simplified manner in a simplified manner.
  • the present invention provides a method for recovering a code word from a received, faulty code word, wherein a code word comprises code units and is derived from one or more information units by means of a redundancy code, with the following steps:
  • step (b) determining a defective or non-existent code unit in the code word; (c) in response to the defective or non-existing code unit determined in step (b), setting a depuncturing pattern;
  • the present invention also provides a device for recovering a code word from a received, incorrect code word, wherein a code word comprises a plurality of code units and was generated by means of a redundancy code
  • a signal processing unit which receives the faulty code word, which determines a faulty or non-existent code unit in the code word, which sets a de-puncturing pattern in response to the faulty or non-existent code unit, and which punctures the received code word based on the set de-puncturing pattern, so that the defective or non-existent code unit is depunctured in the code word.
  • the present invention further provides a method for generating a code word, which contains code units, which are generated based on information units by a redundancy code, for transmission over a lossy connection network, with the following steps:
  • the present invention further provides a device for generating a code word, which contains code units which are generated based on information units by a redundancy code, for transmission via a lossy connection network, comprising:
  • a processing device for arranging the generated code units in the code word such that a loss of one or more code units in the code word leads to a code unit sequence which corresponds to a puncturing pattern applicable to the original code word, which is also known to a recipient of the code word ,
  • a plurality of packets are generated which contain code units which are generated based on information units by a redundancy code and which are transmitted via a packet-oriented connection network.
  • the information units are received and each of the received information units is encoded using a plurality of generators, one information unit being applied to each of the generators and each of the generators generating a code unit for an adjacent information unit.
  • the code units are assigned to a plurality of packets, with code units generated by the same generators being assigned to the same packets. ordered and the generated packages are provided.
  • the present invention also provides a transmission system for transmitting units of information
  • a transmitter which comprises a device for generating a code word or a device which operates according to the inventive method for generating a code word;
  • the receiver which receives the code words transmitted by the transmission medium, the receiver comprising a device for recovering code words according to the present invention, or a device which operates according to the method according to the invention for recovering the code words.
  • the present invention is basically to be assigned to the channel coding method described above and is based on punctured channel codes, which are also described by J. Hagenauer, "Rat-Compatible Punctured Convolutional Codes (RCPC Codes) and their Applications", IEEE Transactions on Communications, Vol. 36 4, pp. 389-400, April 1988.
  • RCPC Codes Random-Compatible Punctured Convolutional Codes
  • the puncturing of the mother code always takes place immediately after the channel encoder on the transmitter side according to a defined puncturing pattern which is known to both the transmitter and the receiver.
  • the present invention leaves this approach and according to the invention there is no puncturing of the channel code on the transmitter side. This means that basically all bits of the generated mother code are sent.
  • the puncturing technique is not used on the transmitter side, but is shifted to the receiver side, the puncturing actually occurring, i.e. the discarding of certain bits, being caused by transmission losses, i.e. by the fact that during transmission a Bitstream single or multiple bits are lost, i.e. they do not arrive at the receiver. This can occur, for example, when transmitting via a packet loss channel.
  • the essence of the present invention is therefore that the puncturing is virtually shifted from the transmitter to the receiver, the actual puncturing being effected by losses in the transmission network.
  • the present invention preferably relates to the use of convolutional codes, since the puncturing technique is particularly suitable for this. However, the present invention is not limited to this.
  • This packet loss channel is considered as an example for the further explanation, although it is obvious that that the present invention is not limited to such a packet loss channel, but can be applied to any lossy transmission channel and the associated transmission methods.
  • the method according to the invention can thus very efficiently correct both packet losses and individual, randomly distributed bit errors in one step.
  • the method according to the invention can be applied to any channel coding method, e.g. B. convolutional codes, turbo codes, etc.
  • a decoder with a soft input or at least one erasure decoder (“erasure”) should preferably be available for the corresponding channel code.
  • the method according to the invention thus leads to a substantially secure transmission of the data in comparison with conventional, commercial methods as described above, which, for example, in the transmission of multimedia data, depending on the application, is significantly improved. voice quality, audio quality, picture quality or video quality.
  • the method according to the invention processes the data before sending it in such a way that a packet loss leads to optimal puncturing of the convolutional code.
  • This is achieved by optimally distributing the encoded data over a plurality of packets. This ensures that a packet loss that would normally not have been correctable by a convolutional code remains correctable up to a certain packet error rate.
  • the preprocessing at the transmitter can guarantee at the receiver that M 'packets sent can be lost, M' ⁇ M, and an error-free decoding result can nevertheless be achieved. M 'depends on the code rate N and the code used.
  • the method according to the invention is used on a combination of fixed network and mobile radio network, but the method according to the invention also works according to another exemplary embodiment for pure packet loss networks or pure mobile radio networks.
  • the use in the combination of the two networks mentioned above is particularly suitable, since here both Packet losses of the fixed network and the bit errors of the mobile radio network can be corrected in one step with the same code, which, as already explained above, was not possible in the prior art.
  • the invention is based on channel codes punctured by packet loss
  • a coding can optionally be used on a base station in a configuration of the transmission network as a combination of a fixed network and a mobile network, without the recipient having to be aware of this,
  • the sender can vary the number and length of packets as desired without the recipient needing to be aware of it, so that the system can be adapted very flexibly to the transmission quality and capacity of the individual subnets
  • the recipient only needs to be informed that a packet has been lost, due to the structure, all adjustments to the system, such as B. mode, rate, carrier, are not signaled to the receiver, so that this adaptation can be done flexibly and depending on the load,
  • Adjustments to the cellular network do not have to be signaled to the transmitter, such as. B. discarding packets because the cellular connection has fewer errors,
  • FIG. 1 shows a flowchart of the method according to the invention in accordance with a preferred exemplary embodiment
  • FIG. 2 shows a transmission system according to a preferred exemplary embodiment of the invention with the receiver according to the invention
  • FIG. 3 shows the behavior of the transmission system shown in FIG. 2 without re-coding the code words received in the base station
  • FIG. 4 shows the behavior of the transmission system shown in FIG. 2 with re-coding of the code words received in the base station; 5 shows a conventional transmission system for packet-oriented transmission between transmitter and receiver;
  • Fig. 6 illustrates the behavior of the conventional transmission system shown in Fig. 5.
  • Fig. 7 is a tabular comparison of the behavior of the conventional transmission system and the transmission system according to the invention.
  • a preferred exemplary embodiment of the method according to the invention for recovering a code word is explained in more detail with reference to FIG. 1.
  • This method assumes that the data stream transmitted from a transmitter to a receiver comprises a plurality of code words, wherein individual code words have a plurality of packets, with a plurality of code units, e.g. B. bits are included, which were generated on the transmitter side by a suitable coding method and were distributed to individual packets according to a predetermined scheme.
  • step S100 the code words are received and in step S102 it is checked whether all packets of the code word have been received. If this is the case, the method goes to step S104, in which the received code word is decoded so as to recover the information units on which the coding in the transmitter was based, that is to say the transmitted multimedia data. Then the decoded data for further processing, eg. B. display, manipulation, etc. forwarded. If it is determined in step S102 that not all packets of the code word have been received, the method proceeds to step S110, in which it is determined for the received code word how many packets have not been received and at which position / position they have been arranged in the code word are.
  • step S112 Based on this information, it is determined in the following step S112 that the received code word, at least from the point of view of the receiver, has obviously been punctured at the transmitter in accordance with a predetermined puncturing pattern. Based on this information, the method according to the invention selects a suitable depuncturing pattern in step S112, which is used in subsequent step S114 to depuncture the code word, as a result of which the code word is completed again. After the code word has been completed, the method goes to step S104, in which the required decoding of the code word can be carried out.
  • the decision S102 made regarding the number of packets received is based on another information signal which indicates whether or not a particular packet from a set of transmitted packets has been received.
  • another information signal which indicates whether or not a particular packet from a set of transmitted packets has been received.
  • a period of time during which the packets of a code word are expected can be monitored, and in the event that one or more packets are received outside this period of time, these are rated as not received.
  • the multimedia data to be transmitted are from various multimedia sources, e.g. B. a camera 100a or a microphone 100b, provided to a multimedia encoder 102.
  • the multimedia encoder 102 generates a plurality of information units based on the signals present from the multimedia sources 100a, 100b, which are forwarded to a channel coding unit 106 via a connection 104.
  • the channel coding unit 106 comprises an FEC encoder 108, a packet distribution unit 110 and a multiplexer 112.
  • the FEC encoder 108 receives the information units at its input via the connection 104 and generates N code units, e.g. B. Bits that are forwarded to the packet distribution unit 110 via the connections shown in FIG. 2.
  • the packet distribution unit 110 distributes the received N code units to M packets, preferably M> N.
  • the M packets are passed on to the multiplexer 112, which then provides them serially at its output.
  • the output of the channel coding unit 106 ie the output of the multiplexer 112, provides the packets generated to a packet loss network, that is to say a packet-oriented connection network 114, which serially transmits the packets to a base station 116.
  • the base station 116 comprises a demultiplexer 118 which provides the M packets transmitted via the network 114 in parallel at an output thereof.
  • a new coding unit 120 can be provided in the base station 116, in which case that at the output of the demultiplexer 118 parallel M packets can be provided in parallel to the re-encoding unit 120.
  • the base station transmits the received M packets, either the M packets received directly at the base station or the M packets newly coded in the base station, via a radio network shown only schematically at 122 in FIG. 2, to an FEC decoder 124 in the mobile receiver.
  • the FEC decoder 124 generates the information units initially provided to the channel coding unit 106 from the received M packets and transmits them via the connection 126 to a multimedia decoder 128 which decodes the received information units and corresponding multimedia devices which are suitable for reproducing the corresponding multimedia data. provides e.g. B. the screen 130a or the speaker 130b.
  • the transmission system shown by way of example in FIG. 2 further comprises a controller 132 for the packet loss network and a controller 134 for the mobile network.
  • DQI Downlink Quality Indication
  • the controller 132 for the fixed network and the controller 134 for the mobile network exchange information which will be discussed later.
  • PER Packet Error Rate
  • the controller for the fixed network 132 issues the channel coding unit 106 with the rate command, which is used to determine the rate at which the packets are sent to the packet loss network 114.
  • the controller for the landline 132 also issues a mode command to the multimedia encoder 102 to determine a mode thereof.
  • provision can be made to provide an optional return channel between the FEC decoder 124 and the channel coding unit 106.
  • FIG. 2 shows the transmission system according to a preferred exemplary embodiment in a possible application scenario of the present invention.
  • the data sources e.g. B. digitized source data provided by the camera 100a or the microphone 100b, processed by the multimedia encoder 102.
  • the original data is compressed.
  • the degree of compression on the multimedia encoder 102 can be adjusted, so that the data rate at the output of the multimedia encoder can be adjusted in several stages by means of the mode command which is present at a control input of the multimedia encoder 102 ,
  • the compressed data is then provided to the channel coding unit 106 via the connection 104.
  • the data is processed by means of the channel encoder or FEC encoder 108, which has a rate of 1 / N.
  • FEC encoder 108 which has a rate of 1 / N.
  • the rate of the FEC encoder 108 is set by means of the rate command which is present at a control input of the FEC encoder 108.
  • the coded bits are now the Packet distribution unit 110 is provided which divides the encoded bits into M packets.
  • the so-called free distance d fr ee which the punctured code has, and its distance spectrum are preferably used to assess puncturing patterns with regard to their “optimality”.
  • the distance spectrum consists of two components ad and Cd.
  • a d gives the number the paths in the trellis that build a Hamming distance d to the null word and C d the sum of the bit errors on these paths of distance d.
  • the optimal puncturing pattern can usually only be determined using complex computer simulations.
  • the puncturing patterns depend on the code, the code rate, the packet length and the number of lost packets. at Convolutional codes are preferred to distribute the punctured bits as evenly as possible over a block.
  • the M packets are sequentially sent over the packet loss channel 114.
  • the network control unit 132 By returning the packet error rate (PER) of the packet loss network 114 to the network control unit 132, it is possible to adaptively set both the compression of the multimedia encoder 102 and the rate of the FEC encoder 108. This makes it possible to adaptively adapt both the number of packets and the length of the packet to the current packet loss probability of the network.
  • PER packet error rate
  • the received data packets are transmitted at the base station (BS) 116 via the carriers of a mobile radio network 122.
  • a control or control channel or in-band signaling can be used for this. The same can be achieved if the packets that have not arrived after a certain time (“time out ”) in the FEC decoder 124 are declared as dotted.
  • decoding with subsequent re-coding 120 can be carried out optionally on the base station 116 before transmission over the radio channel 112.
  • decoding with subsequent re-coding 120 can be carried out optionally on the base station 116 before transmission over the radio channel 112.
  • the packet losses occurring in the packet loss network 114 are corrected before the radio transmission. This becomes a clear gain in particular if the bit error probability on radio channel 122 is high.
  • the received data are then decoded in the FEC decoder 124 at the receiver end and forwarded to the multi-ediadecoder 128, which decompresses them.
  • the method according to the invention can in principle be used in any communication system, although it is particularly suitable for systems in which the transmission characteristic is both packet loss and individual , randomly distributed bit errors in the corresponding data stream.
  • Voice transmission over the Internet and GSM-GPRS, voice / audio / image / video transmission over the Internet and UMTS, voice / audio / image / video transmission over the Internet and GSM are particularly suitable for the present invention.
  • EDGE viewed.
  • data can be transferred very efficiently via the system described above (file transfer).
  • the transmission properties and the behavior of the transmission system described with reference to FIG. 2 according to a preferred implementation of the same are explained in more detail below.
  • the term “mobile Internet” means the combined network described above, consisting of a packet-oriented network part and an additional radio channel connection section.
  • the transmission system opens up a very flexible adaptation of the transmission characteristics depending on the losses on the packet-oriented network and As mentioned, the packet loss at the receiver is considered as puncturing the convolutional code, so the only additional information to be transmitted to the decoder is the bad frame indication mentioned above.
  • the multimedia encoder 102 encodes the speech based on the AMR speech codec, which is based on the CELP technology, which enables up to eight rates between 4.75 kbit / s to 12.2 kbit / s.
  • the subjectively pure voice quality of the 12.2 kbit / s mode is comparable to the G.711 codec.
  • a 6-bit CRC is applied to the most important bits. This CRC is used on the speech decoder for the detection of defective frames of a speech frame and controls the error coverage erroneous language frames. This leads to a total bit rate of 5.05 kbit / s to 12.5 kbit / s after the coding of the speech by the encoder 102.
  • the encoder 102 can also pass on source significance information (SSI) to the channel coding unit 106, which is then used, for example, for the unequal error protection (UEP) of the voice data.
  • SSI source significance information
  • UDP unequal error protection
  • the packet formation is carried out in such a way that all bits which come from a specific generator polynomial G X (D) are assigned to the same packet x. Therefore, at the output of the packet distribution unit 110, there is a packet that contains the systematic bits and Nl different packets, each of which contains the bits that were generated by a specific generator polynomial.
  • the N packets are then serially transmitted over the lossy packet-oriented network 114.
  • the speech coding mode and the rate of the RSC code can be adjusted adaptively, and consequently the number and length of the packets transmitted can be set.
  • the received packets are transmitted at the base station 116 via the carriers i to N N of the wireless network or radio network 122.
  • BFIx bad frame indicator
  • the optional re-encoding described above can be performed at base station 116 prior to transmitting the received packets. This is very helpful in dealing with the loss of packets on the landline 114, which can improve the overall behavior of the system.
  • Feedback of the DQI signal from decoder 124 to the base station via a control unit of the mobile radio network can be used to adaptively adapt the number of carriers used for the wireless connection.
  • the received packets are decoded at the receiver end, that is to say in the decoder 124, and the decoder 128 is designed as a speech decoder which receives the decoded packets. It should be noted that it is sufficient if only one random packet from the N packets, which is received without errors, is required to enable error-free decoding. If any remaining redundancy remains in the speech-encoded data, iterative source and channel decoding can alternatively or additionally be used.
  • the number of packets N was chosen in such a way that the packet distribution is comparable both in the packet according to the invention, ie also in a reference system, which will be described later.
  • the delay of the two systems is also the same.
  • all feedback information and adaptations in the system according to the invention have been neglected for the coding experiment.
  • the AMR mode MR122 was used as the speech encoder, with a fixed rate of 12.2 kbit / s.
  • the 6-bit CRC described above was applied to the most important bits, resulting in a total bit rate of 12.5 kbit / s after the voice coding.
  • the source significance information mentioned above was not used in this example.
  • a code with the rate 1/5 was selected as the recursive systematic convolutional code, with the following generator polynomials:
  • a simple model of a lossy packet-oriented network was assumed for the transmission channel, this simple model assuming statistically independent packet losses, and for the wireless connection additive channels with white distributed noise (AWGN).
  • AWGN white distributed noise
  • S x ( ⁇ ) and S y ( ⁇ ) are the amplitude spectra of the original, error-free transmitted system and the sig- output of the speech decoder.
  • W denotes the signal bandwidth.
  • FIG. 3 shows the behavior of the system described in more detail in FIG. 2 according to the implementation mentioned above, wherein no re-coding was carried out in the base station 116 here.
  • the spectral distance SD is plotted along the z axis as a measure of the voice quality
  • the packet error rate PER occurring on the network 114 is plotted along the x axis
  • the signal-to-noise ratio is plotted on the mobile network along the y axis
  • a reference system is explained in more detail below with reference to FIG. 5, which also transmits encoded voice data to a mobile subscriber via a lossy packet-oriented network and via a wireless connection, but uses a conventional approach to correcting packet losses, namely signal interpolation.
  • an encoder 202 is shown which, via a connection 204, forwards the encoded data to a packet distribution unit 210, which distributes the received data to a multiplicity of packets which are then transmitted to a base station 216 via the network 214.
  • the received packets are transmitted to the mobile receiver via a radio channel 220, the received packets being demultiplexed in the demultiplexer 224 and passed on to the decoder 228.
  • FIG. 5 shows an encoder 202 which, via a connection 204, forwards the encoded data to a packet distribution unit 210, which distributes the received data to a multiplicity of packets which are then transmitted to a base station 216 via the network 214.
  • the received packets are transmitted to the mobile
  • the G.711 codec was chosen as the speech codec and the encoded data stream was distributed over N packets in such a way that adjacent samples were introduced into adjacent packets. Consequently, a packet loss leads to a distribution of the sample signal losses and not to a complete loss of sample values. A packet loss can then be concealed at the decoder simply by interpolating between the distributed sampling losses.
  • the packets are transmitted via the same mobile Internet channel as in the exemplary embodiment of the present invention described above.
  • FIG. 6 shows the behavior of the transmission system shown in FIG. 5, and as can already be seen from a superficial comparison of FIGS. 3 and 4 with FIG. 6, the behavior of the reference system has a significantly poorer speech quality Areas in which the system according to the invention is still used for production language in the transmission. As can be seen from FIG. 6, the system suffers greatly from the non-coded transmission of the voice data via the wireless connection. But even with low bit error rates or high signal-to-noise ratios, the system is still very susceptible to transmission errors as far as the voice quality is concerned. If one now looks at the packet error rate of the lossy packet-oriented network, it can be seen that the reference system is also very sensitive to packet losses.
  • a packet loss rate of 10 -3 already leads to a decrease in the speech quality, wherein, starting from a packet error rate of 10 -3 with increasing packet error rate, the transmission quality is getting worse.
  • the system according to the present invention without re-coding, it can be ascertained that in the reference system, even with an increasing packet error rate from 10 "3, a significant decrease in the speech quality can be determined, which does not occur in the system according to the invention.
  • the present invention can also be applied to systems in which, in order to increase the transmission capacity, the generated codes are punctured, with an additional loss in the transmission being regarded as additional puncturing by the receiver.
  • the loss of the packets or the code units can also be caused in the receiver and / or after generation in the transmitter, in addition to the transmission.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Detection And Prevention Of Errors In Transmission (AREA)
  • Error Detection And Correction (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

L'invention concerne un procédé et un dispositif permettant de récupérer un mot de code à partir d'un mot de code erroné reçu. Selon ce procédé, le mot de code erroné est d'abord reçu (5100). Le mot de code comprend des unités de code et a été dérivé, à l'aide d'un code de redondance d'une ou de plusieurs unités d'information. Après réception du mot de code erroné, les unités de code erronées ou non présentes sont déterminées dans le mot de code (5110), puis un modèle de déponctuation est adapté auxdites unités, de manière correspondante. Sur la base du modèle de déponctuation adapté, le mot de code erroné reçu est alors déponctué, de sorte que l'unité de code erronée ou non présente soit déponctuée dans le mot de code.
PCT/EP2003/004758 2002-05-07 2003-05-06 Procede et dispositif pour recuperer un mot de code a partir d'un mot de code errone reçu et dispositif pour produire un mot de code, et systeme de controle Ceased WO2003096538A2 (fr)

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AU2003226827A AU2003226827A1 (en) 2002-05-07 2003-05-06 Method and device for recovering a code word from a received faulty code word, method and device for generating a code word, and transmission system

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DE2002120370 DE10220370A1 (de) 2002-05-07 2002-05-07 Verfahren und Vorrichtung zum Wiedergewinnen eines Codeworts aus einem empfangenen, fehlerhaften Codewort, Verfahren und Vorrichtung zum Erzeugen eines Codeworts, und Übertragungssystem
DE10220370.9 2002-05-07

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WO2003096538A3 WO2003096538A3 (fr) 2004-04-08

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