Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/16—Vocoder architecture
  • G10L19/173—Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding

Definitions

• the present invention relates to data processing by passing between different subband domains, including, but not exclusively, for transcoding between two types of compression coding / decoding.
• One of the main problems due to the heterogeneity of terminals concerns the diversity of coding formats that they are able to interpret.
• One possible solution would be to recover the capabilities of the terminal before delivering the content in a compatible format. This solution may be more or less effective depending on the delivery scenario of the multimedia content considered
• transcoding or changing the encoding format
• This operation can take place at different levels of the transmission chain. It can intervene at the server to change the format of content previously stored for example in a database, or intervene in a gateway in the network, or other.
• a common and straightforward method of transcoding is to decode the content and recode it to obtain a representation in the new encoding format.
• This method generally has the drawbacks of using a large computing power, of increasing the algorithmic delay due to the processing and sometimes of adding further degradation of the perceptual quality of the multimedia signal. These settings are very important in multimedia applications. Their improvement
• This type of transcoding consists in performing a partial decoding, as minimal as possible, of the initial coding format to extract the parameters allowing the reconstruction of the new coding format.
• the success of this process It is therefore able to reduce algorithmic complexity and delay and to maintain or even increase perceptual quality.
• audio transcoding is a definition of audio transcoding and the main problems that arise after a brief review of the principles of perceptual audio coding in subbands.
• audio coders can be manufacturer-specific (or "proprietary"), or standardized by decision of international organizations. In addition, they all have a common basic structure and are based on the same principles.
• the basic principle of perceptual frequency audio coding is to reduce the flow of information by exploiting the properties of the human hearing system.
• the irrelevant components of the audio signal are eliminated.
• This operation uses the phenomenon called "masking". Since the description of this masking effect is mainly in the frequency domain, the representation of the signal is carried out in the frequency domain.
• FIG. 2a the input digital audio signal Se is first decomposed by an analysis filter bank 20.
• the resulting spectral components are then quantized and then coded by the module 22.
• Quantization uses the result of a perceptual model 24 so that the noise that results from the treatment is inaudible.
• a multiplexing of the different coded parameters is performed by the module 26 and an audio frame Sc is thus constructed.
• the decoding is done in a dual manner. After demultiplexing the audio frame by the module 21, the various parameters are decoded and the spectral components of the signal are de-quantized by the module 23.
• the temporal audio signal is reconstituted by the synthesis filterbank 25.
• the first stage of any perceptual audio coding system therefore consists of an analysis filter bank 20 used for the time / frequency transformation.
• filter banks and transforms have been developed and exploited in audio coders.
• pseudo-QMF filter banks, hybrid filter banks, and MDCT transform banks can be mentioned.
• the MDCT transform is currently proving to be the most effective in this context. It is the basis of the latest and most advanced audio coding algorithms such as those used for MPEG-4 AAC, TwinVQ and BSAC, Dolby AC-3, in the TDAC encoder / decoder (for "rhyme Domain Aliasing Canceling"). ) of France Telecom, in ITU-T G.722.1.
• Modulated cosine filter banks orthogonal transforms (or "LOT" for "Lapped Orthogonal Transform ") and more generally for the banks of maximum decimation filters, that is to say to critical sampling.
• Critical sampling property for a filter bank is that the subsampling / oversampling factor is equal to the number of subbands.
• FIGS. 3a and 3b respectively illustrate the conventional transcoding and intelligent transcoding schemes in a communication chain, between a coder CO1 according to a first coding format and a decoder DEC2 according to a second coding format.
• conventional transcoding it is a question of carrying out a complete decoding operation by the decoder module DECl according to the first format (FIG. 3a), followed by recoding by the encoder module C02 according to the second format, to finally arrive at the second coding format.
• FIG. 4 shows the details of the operations that are merged by the implementation of intelligent transcoding. It mainly involves integrating the functional blocks of the synthesis filter banks BS1 and the BA2 analysis filter banks of the conventional transcoding into a system for direct conversion between subband domains in the module 31.
• Table 1 gives a summary of the types of filter banks used in the most well-known transform audio coders, as well as their characteristics. As can be seen, in addition to the MDCT transform which is the most used, there are Pseudo-QMF benches. Moreover, they are all part of the family of maximum decimation banks and modulated cosines verifying exactly or almost the perfect reconstruction property.
• Table 1 The most used filter banks in audio coding and their characteristics.
• Table 2 below shows some types of subband coding in Table 1, detailing some of their applications.
• Table 2 Examples of subband coders for audio and speech signals and some examples of their main applications.
• US-6,134,523 discloses a rate reduction method in the coded domain for audio coded in MPEG-I Layer I or II signals. Although this method is similar to audio transcoding methods, it does not make any change between coding formats and the signals of the subbands remain in the representation of the same transformed domain, namely the representation of the pseudo filter bank. QMF. Here, the signals are simply re-quantized according to a new bit allocation. Moreover, in US-2003/0149559, a method is proposed for reducing the complexity of the psycho ⁇ acoustic model during a transcoding operation.
• the new system uses values stored in a database of distortion jigs. Even if this method deals with a problem of transcoding, it remains far from the objectives relative to the passage between filter bank domains.
• This technique of the prior art can be applied only for this particular case of transcoding. • This technique does not really treat a conversion to a different new subband field. It's just a matter of cascading a new Missing analysis filter bank, which allows to increase the frequency resolution.
• TDF Tranform Domain Filtering
• TDRT Transform-Domain Resolution Translation
• DCT for "Discrete Cosine Transform”
• MLT Modulated Lapped Transform
• This publication discloses an efficient structure for implementing a synthesis filter bank system, at L sub-bands, followed by an M sub-band analysis filter bank, where M and L are multiple. one of the other.
• This structure is effective for implementation in VLSI ("Very Large Scale Integration") or FPGA ("Field Programmable Gate Array”) or parallel processors. It requires fewer logic blocks, low power consumption and allows the degree of parallelism to be extended.
• the proposed method is applicable in situations where subband-based processing follows another subband treatment and where the synthesized intermediate signal is not needed.
• TDM to FDM trans-multiplexing for "Time Domain Multiplexing” to "Frequency Domain Multiplexing"
• a synthesis filter bank is used to reconstruct the interleaved time signals (that is to say perform the operation of reverse multiplexing from FDM to TDM).
• an analysis filter bank is used to reconstruct the interleaved time signals (that is to say perform the operation of reverse multiplexing from FDM to TDM).
• the structure of the TDM->FDM-> TDM system thus amounts to a cascading of a synthesis filter bank and an analysis filter bank, which corresponds to what is also used in a system. conventional transcoding.
• the problem generally posed in these trans-multiplexing systems is to reconstruct the original signals without distortions after the TDM->FDM-> TDM operation.
• the present invention improves the situation with respect to the state of the art presented above.
• the present invention proposes in particular, but not exclusively, as will be seen below, a transcoding of a first type of coding, any, to a second type of coding, any.
• the respective numbers of M and Z sub-bands are any natural numbers and are not necessarily linked by a proportionality relation, in the most general case.
• the method in the sense of the invention can advantageously be applied to the transcoding of a first type of encoding / decoding in compression to at least a second type of encoding / decoding in compression.
• This application typically consists in compacting in the same treatment the following steps:
• the present invention also relates to a computer program product, intended to be stored in a memory of a device in a communication network, such as a server, a gateway, or a terminal, and then including instructions for the implementation of of all or part of the process according to the invention.
• the present invention also relates to equipment such as a server, a gateway, or a terminal for a communication network, and comprising computer resources for implementing the method according to the invention.
• FIGS. 2a and 2b showing the block diagrams of a perceptual frequency audio compression system, respectively to coding and decoding
• FIGS. 3a and 3b schematically illustrating communication channels using transcoding. conventional and intelligent transcoding, respectively, and
• FIG. 4 represents the block diagrams illustrating the conventional transcoding (upper part of the figure) and the intelligent transcoding (lower part of the figure), described above,
• FIGS. 5a and 5b schematically represent the block diagrams defining the equivalence between the synthesis of the temporal signal and then the analysis with a new bank of filters (FIG. 5a) and the direct conversion between two domains of the subbands (FIG. 5b).
• FIG. 6 illustrates a representation in multi -addition blocks of the conventional conversion between subband domains
• FIG. 7 is a multi-layer representation of the sub-domain domain conversion system, in the sense of the invention
• FIG. 8 schematically summarizes the filtering method in a conversion system, within the meaning of 1'invention
• FIG. 14 is a representation of the conversion system in the case M ⁇ pL as an LPTV system, with an input rate different from the output rate
• FIG. 15 is a representation of the conversion system within the meaning of the invention, as an LPTV system, in the general case where M and L are not linked by a particular relationship of proportionality,
• FIG. 18 illustrates the conversion system. within the meaning of the invention in an embodiment corresponding to an OLA recovery transform and addition for an efficient implementation allowing on-the-fly processing, in the particular case MPL
• FIG. conversion in the sense of the invention in an embodiment corresponding to a transformation and an addition with OLA overlay for efficient implementation allowing on-the-fly processing in the case by Particularly, FIG. 20a and 20b respectively illustrate a combination filtering with a conversion between domains of FIG. sub-bands, and an equivalent overall system, within the meaning of the invention
• FIGS. 21a and 21b illustrate the combination of a sampling frequency change (or "resampling") with a conversion between subband domains, conventional and in the sense of the invention, respectively
• FIG. is a representation in multiple-frame blocks of the conversion system within the meaning of the invention between subband domains combined with re-sampling
• FIG. 23 represents the system within the meaning of the invention as an LPTV system applied to a combined conversion with a re-sampling
• FIG. 24 represents a preferred embodiment corresponding to an OLA recovery transform and addition for efficient implementation allowing on-the-fly processing of the conversion system of FIG. 23
• FIG. 25 represents a transcoding occurring in a gateway GW of a communication network, for a possible application of the present invention
• - 27 is a table showing the parameters of the conversion system within the meaning of the invention for particular cases of encoding formats.
• the method of converting between subband domains is described below in a general discussion of the invention.
• the L-band synthesis bench used by a first compression coding system and defined by its filters, denoted by F ⁇ . (Z), O ⁇ k ⁇ LX, and the M band analysis filter bank are considered.
• F ⁇ . (Z) The L-band synthesis bench used by a first compression coding system and defined by its filters, denoted by F ⁇ . (Z), O ⁇ k ⁇ LX, and the M band analysis filter bank are considered.
• a second compression system and defined by its filters, noted
• the signal vectors of the subbands representing the signal respectively in the areas of the first and second bank of filters.
• FIG. 5b The principle of conversion between domains of the subbands is illustrated by Figures 5a and 5b. It is a question of finding a conversion system 51 (FIG. 5b) between the vectors of the subband signals, X ( ⁇ ) and Y ( ⁇ ), equivalent to a cascading of the synthesis bench BS1 and the bank of BA2 analysis (FIG. 5a).
• the objective is to merge certain mathematical calculation operations between these two banks of filters to reduce the algorithmic complexity (that is to say the number of calculation operations and the required memory). Another objective is therefore to minimize the algorithmic delay introduced by this transformation.
• FIG. 5a By using multi-layer blocks, the diagram of FIG. 5a can be represented by that of FIG. 6, on which an analysis filter bank follows a synthesis filter bank.
• the synthesis filter bank subbands Z is conventionally compound in each subband k r O ⁇ k ⁇ Li, an upsampling operation by "a factor L followed by a filtering synthesis filter
• the subband signal corresponding to the kth component of the input vector X (z) is therefore first oversampled and then filtered by the filter F 1 ⁇ z).
• X (z) synthesized at the output of this synthesis bank is then obtained by summing the results of these filterings for Q ⁇ k ⁇ Ll.
• This time signal then constitutes the input of the analysis bank to M subbands. It undergoes on each sub-band n, 0 ⁇ n ⁇ Ml, a filtering by the analysis filter, H M (z), followed by a subsampling operation of factor M. It then obtains at the output of this bench analysis a vector of sub-band signals, size M, shown in the domain of the z-transform Y (z) • synthesis of a time signal is therefore generally necessary in this conventional conversion system, unlike to the conversion system within the meaning of the invention which is described below.
• V (z) T (z) U (z) (4)
• the conversion matrix T (z) is of size KxK. Its expression is given by: (5) where v (z) is the matrix of size p ⁇ xp 2 whose elements are defined as follows:
• the operation ® designates the Kronecker product such as:
• ⁇ K denotes the decimation by a factor K, corresponding to a subsampling where only one sample is selected from K samples.
• the conversion system can be schematized as shown in FIG. 7, which shows that the system is advantageously a so-called “linear periodically variable time” (LPTV) system, as will be seen later.
• LPTV linear periodically variable time
• the input block 71 consisting of the advance P2 ⁇ x and the delay chain, followed by the decimation 72_p 2 -l to 72_0 by a factor p 2 , can be interpreted as a mechanism of blocking each succession of p 2 input vectors, denoted by X ["], into a single vector U [fc], of size K.
• This latter vector U [A:] is then applied to the filtering matrix T (z) (module 74) and the result is a vector V [A:], of the same size as the vector U [A:].
• the notation X ()) simply relates to the expression of the vector X according to its transform in z, while the notation X ["] relates to the expression of the vector X in the time domain, conventionally for the skilled person.
• the last block 73_pi-1 to 73_0 of FIG. 7 finally makes it possible to put in series the successive p ⁇ sub-vectors, each of size M, of the vector V [A:] to have as output the vectors Y [V].
• FIG. 7 The input and output blocks of FIG. 7 are finally little different from the blocking mechanisms 81 and then series-linking mechanisms 82, respectively, of FIG. 8 which summarizes the main steps of the method within the meaning of the invention.
• the conversion system within the meaning of the invention is minimal delay.
• the element filters of the matrix T (z), are all causal if and only if: e mm ⁇ K- ⁇ , - (11)
• Conversion systems within the meaning of the invention can therefore be constructed with different delays and by making different choices on the parameters a and b, but provided that the inequality (12) is preferentially satisfied.
• the parameters a and b can therefore be seen as setting parameters for acting on the algorithmic delay introduced by the conversion system between subband domains.
• v (z) is the matrix whose elements are defined as follows: (17)
• the relation (16) is therefore the general formula of the conversion matrix T (z), which makes it possible to minimize the algorithmic delay introduced by the conversion system within the meaning of the invention.
• polyphase components considered in relation (18) correspond to a type decomposition 1 to the order K 1 as described for example in the aforementioned reference:
• the polyphase components G r n! C (z) (with Q ⁇ r ⁇ Kl) can be determined directly if the synthesis filters and the analysis filters have finite impulse responses (or "FIR"). In the case where one or both banks of filters use recursive filters (with infinite impulse responses or 11 IIR "), the produced filters G nt (z) are also infinite impulse responses.
• the general procedure for such a decomposition is given in Annex A, "Polyphase decomposition of recursive filters", reference:
• the conversion matrix in this case is of size MxM and is written as follows:
• This matrix is therefore the line vector consisting respectively of polyphase components of general index (pk) L1 (where O ⁇ k ⁇ pl), following a type 1 decomposition to order M, of the matrix g (z ), synthesis filter products and analysis.
• the notation G r mJ (z) (with O ⁇ r ⁇ M-1) refers to the polyphase component of general index r of the filter G mJ (z), resulting from a decomposition to the order M.
• the conversion matrix in this case is of size LxL and is written as follows:
• This matrix is therefore the column vector consisting respectively of polyphase components of general index (k + 1) M1 (with O ⁇ k ⁇ p-1), following a type 1 decomposition at order L, of the matrix g ( z), synthesis filter products and analysis.
• the notation G r tl (z) (with O ⁇ r ⁇ X-1) indicates the polyphase component of general index r of the filter G ⁇ z), resulting from a decomposition to the order L.
• FIG. 11 The diagram of the conversion system is given in this case in FIG. 11 in multi-layer representation and in FIG. 12 illustrating the main steps of the filtering method in this particular case where L ⁇ pM.
• This conversion system can be seen as a system
• the input rate of this system is f -pf s and the output rate is f ⁇ .
• the transfer matrices A k (z) operate at the sampling frequency / et and the system operates globally as if a switch 140 (FIG. 14), at the input of the system, was flipping in a circular manner to this same frequency f s , from one input of a matrix block A A (z) to the other.
• the output of the conversion system Y [ w ] 'at the instant nT Si is equal to the sum of the outputs of the ⁇ ⁇ z ) (with O ⁇ k ⁇ p-1), each fed by, at the respective moments:
• the two switches 151 and 152 shown respectively at the input and the output of the structure of FIG. 15 operate with a frequency - which is also
• the input rate of this system is / ⁇ and the output rate is f s , allowing processing of the input data, on the fly, by the conversion system within the meaning of the invention.
• the conversion matrix T (z) is expressed as follows:
• P n are matrices of size KxK, and N corresponds to the maximum of the lengths of the filters T ffl / (z), elements of T (z).
• the system can therefore be constructed by a matrix transform P, followed by a recovery addition operation.
• This implementation is similar to the synthesis part of an overlapping transform "LT" (for "Lapped Transform"), as described in particular in: “Signal Processing with Lapped Transform", HS Malvar, Artech House, Inc. 1992 .
• the method presented below provides a parallelism in the treatment and efficient use of IT resources (-logi-cial-matéxiis ⁇ re or ⁇ s) for the implementation of the process. It is therefore a presently preferred embodiment at least in the case of finite impulse response filter banks.
• each transfer matrix A , (z) contains filters of identical lengths and which depend on the value of e ij , then the corresponding matrix B (J) also depends on e ij.
• the matrices B / contain zero sub-matrices. and their forms are given as follows: o If 0 ⁇ e ij ⁇ r 0 -l then:
• the null blocks of the matrices B / y allow a reduction of calculation during a transformation of an input vector by this matrix.
• Addition with cover 2.c step is done on vectors • NM length with a covering (NI) M elements.
• the output Y ["] of the conversion system corresponds to the result of the overlap addition on the sum vectors resulting from step 3.
• This matrix has the following form:
• Each new input vector X [&] is oriented to the common memory of all the subsystems characterized by the transfer matrices A,. (Z), with 0 ⁇ i ⁇ p-1.
• the filter bank is characterized by the fact that the analysis and synthesis filters are obtained by a cosine modulation of a low-pass protector filter.
• Equations (57), (58) and the above conditions make it possible to fully characterize a modulated cosine filter bank with perfect reconstruction.
• modulated cosine filter banks with perfect reconstruction are the basis of all the filter banks of the current audio coders. Even the pseudo-QMF filter bank of the MPEG-1/2 layer I & II coders can be associated with this category, it being understood that the prototype filter is sufficiently well designed to consider that the perfect reconstruction is satisfied.
• the latter can be considered as an MLT transform (for "Modulated Lapped Transform") also known as MDCT (for "Modified DCT”).
• MDCT Modulated DCT
• This transform is used in most coders current frequency audio (MPEG-2/4 AAC, PAC, MSAudio, TDAC, etc.).
• the window must check the condition of symmetry: and complementarity in power: .
• This window choice is used in TDAC and G.722.1 encoders.
• Another choice is to take a window derived from the Kaiser-Bessel window (or "KBD") as in the case of MPEG-4 AAC, BSAC, Twin VQ and AC-3 encoders.
• the values provided in the MPEG-I Audio Layer I-II standard correspond to the window (-1) h (2lM + j), with 0 ⁇ j ⁇ 2M ⁇ l and 0 ⁇ / ⁇ m-1.
• HRTF filters Head Related Trasfert Functions
• FIG. 5a With respect to the block diagram of FIG. 5a, it is a matter of introducing a filter S (z) between the two banks of synthesis and analysis filters and of finding an equivalent system.
• the block diagrams are shown in FIGS. 20a and 20b.
• the conversion system combined with the filtering can be modeled by the same type of scheme as that shown in FIG. 5b. However, it is characterized by the new filter matrix T (z) defined by:
• the filter S FB (z) is a low pass filter of standardized cutoff frequency and gain in bandwidth Q.
• the conversion system combined with the sampling rate change can be modeled by the scheme of Figure 22. It is characterized by the filter matrix T (z) of size q x Mx.q 2 L, defined as follows:
• g (z) is the matrix of size MxL whose elements are given by: (69) and v (z) is the matrix whose elements are defined as follows: (70) also respecting the following relation: (71)
• G nk [z) is interpreted as the result of the convolution of the filter H n (z) oversampled by a factor R, the filter S P ⁇ (z) and the filter F 4 (z ) oversampled by a Q factor.
• the system according to FIG. 23 operates with the matrices A / y (2) such that:
• matrices B 011 are of size MxL
• the following definition of matrices B 1 as shown in FIG. 24 can be given as follows:
• the present invention provides a generic solution for converting a representation of a signal from one subband (or transform) domain to another.
• the method is preferably applied in the context where the banks of filters used by the two compression systems are maximum decimation, as has been seen above.
• the described embodiments may be provided for all transform or subband coders of multimedia signals, especially those used in video, picture, speech coding, or other.
• These embodiments can also be implemented in any device having a cascade of a synthesis bench and an analysis bench, in particular in the following examples: • Improvement of the quality of the speech in sub-bands followed echo cancellation in sub-bands and vice versa.
• transcoding can occur at different points in the transmission chain. In the following, we distinguish some possible case.
• the transcoding mechanism TRANS is advantageous in a gateway GW in the network RES of transmission of the audio content coming from a server SER and destined for a first terminal TER1, equipped with a decoder DECl and another terminal TER2. equipped with another decoder DEC2, as shown in Figure 25.
• transcoding TRANS FOG. 26
• Terminal capacity information was previously received and analyzed by the SER server.
• the audio content is stored in a given encoding format. It is transcoded in real time to be compatible with the terminal at every request of a user before being downloaded.
• the terminals involved may have different capabilities in terms of coders / decoders.
• transcoding can occur at the bridge.
• Table 3 below now shows some possible transcoding, advantageous, between audio coding formats according to the fields of application.
• Table 3 Examples of some interesting types of transcodings and their areas of application.
• FIG. 27 then indicates the parameters of the conversion system within the meaning of the invention for these particular cases of coding formats.

Landscapes

• Engineering & Computer Science (AREA)
• Physics & Mathematics (AREA)
• Computational Linguistics (AREA)
• Signal Processing (AREA)
• Health & Medical Sciences (AREA)
• Audiology, Speech & Language Pathology (AREA)
• Human Computer Interaction (AREA)
• Acoustics & Sound (AREA)
• Multimedia (AREA)
• Spectroscopy & Molecular Physics (AREA)
• Compression, Expansion, Code Conversion, And Decoders (AREA)
• Image Processing (AREA)

Applications Claiming Priority (2)

Publications (2)

Family

ID=34951876

Family Applications (1)

Country Status (8)

Families Citing this family (25)

Family Cites Families (14)

• 2004
  • 2004-09-16 FR FR0409820A patent/FR2875351A1/fr active Pending
• 2005
  • 2005-08-23 JP JP2007531786A patent/JP4850837B2/ja not_active Expired - Fee Related
  • 2005-08-23 CN CN200580034793.8A patent/CN101069233B/zh not_active Expired - Fee Related
  • 2005-08-23 AT AT05798240T patent/ATE458242T1/de not_active IP Right Cessation
  • 2005-08-23 WO PCT/FR2005/002127 patent/WO2006032740A1/fr active Application Filing
  • 2005-08-23 EP EP05798240A patent/EP1794748B1/de not_active Not-in-force
  • 2005-08-23 US US11/662,980 patent/US8639735B2/en not_active Expired - Fee Related
  • 2005-08-23 DE DE602005019431T patent/DE602005019431D1/de active Active

Non-Patent Citations (1)

Also Published As

Similar Documents

Legal Events