EP1388146A2 - Method for encoding and transmitting voice signals - Google Patents
Method for encoding and transmitting voice signalsInfo
- Publication number
- EP1388146A2 EP1388146A2 EP02740316A EP02740316A EP1388146A2 EP 1388146 A2 EP1388146 A2 EP 1388146A2 EP 02740316 A EP02740316 A EP 02740316A EP 02740316 A EP02740316 A EP 02740316A EP 1388146 A2 EP1388146 A2 EP 1388146A2
- Authority
- EP
- European Patent Office
- Prior art keywords
- gain factor
- adaptive
- speech
- code book
- signal
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
- 238000000034 method Methods 0.000 title claims abstract description 24
- 230000003044 adaptive effect Effects 0.000 claims abstract description 61
- 230000005284 excitation Effects 0.000 claims description 43
- 238000003786 synthesis reaction Methods 0.000 claims description 17
- 230000015572 biosynthetic process Effects 0.000 claims description 14
- 230000003321 amplification Effects 0.000 abstract description 10
- 238000003199 nucleic acid amplification method Methods 0.000 abstract description 10
- 238000013139 quantization Methods 0.000 description 23
- 239000013598 vector Substances 0.000 description 10
- 230000005540 biological transmission Effects 0.000 description 5
- 238000011161 development Methods 0.000 description 5
- 230000018109 developmental process Effects 0.000 description 5
- 230000008901 benefit Effects 0.000 description 4
- 230000000737 periodic effect Effects 0.000 description 4
- 238000001914 filtration Methods 0.000 description 3
- 230000007774 longterm Effects 0.000 description 2
- 238000001208 nuclear magnetic resonance pulse sequence Methods 0.000 description 2
- 241001620634 Roger Species 0.000 description 1
- 230000006978 adaptation Effects 0.000 description 1
- 238000007906 compression Methods 0.000 description 1
- 230000006835 compression Effects 0.000 description 1
- 238000013144 data compression Methods 0.000 description 1
- 230000009849 deactivation Effects 0.000 description 1
- 230000003111 delayed effect Effects 0.000 description 1
- 230000001419 dependent effect Effects 0.000 description 1
- 230000003595 spectral effect Effects 0.000 description 1
- 230000002123 temporal effect Effects 0.000 description 1
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
- G10L19/04—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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain
-
- 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
- G10L19/18—Vocoders using multiple modes
Definitions
- the invention relates to a method for coding voice signals, in particular with the inclusion of several code books, the entries of which are used to approximate the voice signal, and a method for transmitting voice signals.
- voice coding methods are used in order to reduce the bit rate to be transmitted.
- the speech coding methods usually deliver a bit stream of speech-coded bits, which is divided into frames, each representing, for example, 20 ms of the speech signal.
- the bits within a frame generally represent a certain set of parameters.
- a frame in turn is often divided into subframes, so that some parameters are transmitted once per frame, others once per subframe.
- the US-TDMA Enhanced Full Rate (EFR) speech codec with 7.4 kbps is given as an example, ie 148 bits per 20 ms frame.
- a frame consists of 4 subframes.
- CELP coders code-excited linear prediction
- LPC synthesis filter linear predictive coding
- the filter represents the spectral envelope of the speech signal in the area of the current frame.
- the excitation signal for this filter is additively composed of a so-called “adaptive excitation signal” S_a weighted with a so-called “adaptive gain factor” g_l and one with a so-called “fixed Gain factor "g_2 weighted so-called” fixed excitation signal "S_f together.
- the fixed excitation S_f consists of an entry of the so-called “fixed code book”, which is weighted with the fixed gain factor g_2.
- the entries in the fixed code book each consist of a pulse sequence that only differs from zero at a few points in time.
- the adaptive excitation signal in the so-called analysis-by-synthesis CELP coding method is determined from the excitation signal of the LPC synthesis filter, delayed by a period of the basic speech frequency. All possible quantized basic speech frequencies constitute the so-called "adaptive code book", which contains the correspondingly shifted excitation signals.
- the entries in a code book are generally called code words or code vectors.
- the adaptive code book is called “adaptive” because the code vectors contained in it do not represent constants or are even stored, but instead are determined adaptively for each subframe from the past of the total excitation signal of the LPC synthesis filter.
- the fixed code book is "fixed” insofar as its code vectors are either stored permanently (noise excitation) or are at least calculated using determined computing steps (algebraic code book) that are not dependent on the respective subframe are.
- the respective assigned amplification factors are usually also referred to as “adaptive” or "fixed”. It should be noted that all 4 parameter types, adaptive and fixed excitation signal, as well as adaptive and fixed amplification factor, are of course to be determined in each subframe, and in this sense all are “adaptive in nature". In the following, however, the terminology previously introduced - which is also common in the literature - should be adhered to or the term “first gain factor” should be used instead of “adaptive gain factor” and the term second gain factor should be used instead of "fixed gain factor”.
- the excitation signal S v should reflect as precisely as possible the speech section occurring at that time, the speech signal S.
- the parameters g_l, g_2, S_a, S_f are therefore chosen so that the speech signal S can be represented as well as possible.
- the excitation signal S v g_l * S_a + g_2 * S_f thus approximates the speech signal after LPC synthesis filtering on the receiver side.
- Speech signals contain sequences of frames or subframes in which they can be modeled as stationary, i.e. without the temporal development of their statistical properties. These are periodic sections that can represent vowels, for example. This periodicity flows into the entire excitation signal S ⁇ via the contribution g_l * S_a. However, there are also deeply non-stationary speech signal sections, such as so-called "onsets” or "speech onsets”. These are, for example, plosive sounds at the beginning of a word. In this case, the mand g_2 * S_f represents the dominant contribution to the excitation signal S x .
- the statistical properties of a frame or subframe with an onset cannot usually be estimated from past frames or subframes.
- no long-term periodicity can be determined, that is to say the value of a basic speech frequency is completely meaningless and useless.
- the contribution made up of the adaptive gain factor and entry of the adaptive code book, which expresses long-term periodicity in the speech signal, is therefore more of a hindrance than onsets for coding the speech signal section.
- the contribution of an adaptive excitation signal to the overall excitation signal in onsets can actually hurt: If there is no periodicity at all, that is, no suitable adaptive excitation signal in the context of the adaptive code book search, the optimal adaptive gain factor is zero.
- Adaptive and fixed gain factors g_l and g_2 are now often quantized as a pair of numbers (g_l, g_2) by means of a further code book for the gain factors.
- Scalar quantization is an individual, independent quantization of the parameters Roger that. As already mentioned above, the number of entries in this code book is limited.
- Some speech coders such as the GSM Enhanced Full Rate Coder (GSM-EFR), perform scalar quantization of the gain factors.
- GSM-EFR GSM Enhanced Full Rate Coder
- the adaptive gain factor with 4 bits per subframe and the fixed gain factor with 5 bits per subframe are quantized individually and independently of one another.
- This has the advantage that with certain non-stationary languages cut, for example in the onsets, the adaptive gain factor can easily be quantized to zero, and the fixed gain factor can assume an independent value after quantization.
- GSM-HR GSM half-rate coder
- the present invention is therefore based on the object of specifying a method for coding and for transmission which works in a space-saving manner, works efficiently and is not prone to errors, in particular is efficient in terms of complexity and coding and at the same time has a high signal quality after decoding.
- the value of the first amplification factor which is assigned to an adaptive code book, is set for certain values of a signal classifier.
- the speech signal is broken down into individual time segments. These sections can represent frames (frames) or subframes (subframes), for example.
- the signal classifier indicates, for example, whether there is a stationary or a non-stationary speech section, that is to say whether it is a speech onset, for example. If such a case now exists, a value determined by the signal classifier can be assigned to the first amplification factor.
- this value of the first gain factor can be set such that this representation of the value requires fewer bits than a conventional representation.
- this method proves to be advantageous if the first gain factor is set to zero. This increases the quality of the speech-decoded signal, since, as stated at the beginning, fewer quantization error signal components occur in non-stationary speech sections, for example.
- the second gain factor is scalarly quantized if the first gain factor is fixed. For example, the Resolution of the quantization of the second gain factor can be increased.
- the encoder operates at a fixed data rate, that is to say that a fixed amount of data is provided for a section of a speech signal.
- the reduction in the amount of data achieved to represent the first gain factor and, alternatively or optionally, the adaptive codebook entry can be exploited to the extent that the portion of the amount of data which is now not occupied by data is used to represent other parameters which occur during speech coding.
- Another development provides that the speech section is displayed with a reduced amount of data. This method can be used in particular when using a coding method with a variable bit rate.
- the invention relates to a method for the transmission of voice signals which are encoded according to one of the preceding claims. It is important here that the first gain factor and / or the adaptive codebook entry is not transmitted.
- this method has advantages if the
- Receiver for example the decoder
- Receiver is indicated by information that this reduction in the amount of data was carried out to represent individual parameters.
- This information can, for example, occupy a portion of the amount of data not occupied by the reduction or can also be sent in addition to the amount of data of the frame or subframe.
- Figure 1 shows an overview of the analysis-by-synthesis principle in speech coding
- Figure 2 shows the use of adaptive and fixed code book with the associated gain factors.
- Figure 1 shows the schematic flow of a speech coding according to the analysis-by-synthesis principle.
- the original speech signal 10 is compared with a synthesized speech signal 11.
- the synthesized speech signal 11 should be such that the deviation between the synthesized speech signal 11 and the original speech signal 10 is minimal. If necessary, this deviation is also spectrally weighted. This is done via a weighting filter W (z).
- the synthesized speech signal is produced using an LPC synthesis filter H (z). This synthesis filter is excited via an excitation signal 12. The parameters of this excitation signal 12 (and possibly also the coefficients of the LPC synthesis filter) are ultimately transmitted and should therefore be coded as efficiently as possible.
- FIG. 2 shows the excitation generator in detail without a downstream LPC synthesis filter.
- the excitation signal 12 is composed of an adaptive part, by means of which periodic speech sections are predominantly represented, and a fixed part, which serves to represent non-periodic sections. This has already been explained in detail at the beginning.
- the entries in the adaptive codebook 1 are determined by the preceding language sections. This is done via a feedback loop 2.
- the first gain factor 3 is determined by adapting to the original speech signal 10.
- the fixed code book 4 contains entries which are not determined by a previous period.
- Each entry in the code book, the so-called code word, an algebraic code vector is a pulse sequence that only has non-zero values at a few defined points in time.
- This entry or excitation sequence is selected, by means of which the deviation of the synthesized signal 11 from the original speech signal 10 is minimized.
- the gain factor 5 assigned to the fixed codebook is determined accordingly.
- a so-called signal classifier is calculated for each frame.
- This signal classifier can, for example, provide a binary decision as to whether the adaptive code book should be used or not. For this purpose, it can be an onset recognizer. It is provided that, depending on the classifier, the adaptive gain factor is set to zero, that is to say the adaptive excitation is not included in the overall excitation signal of the LPC synthesis filter. It is also provided that at least one parameter is no longer transmitted. There are several sensible alternatives for this:
- the adaptive codebook entry (that is to say the fundamental speech frequency) no longer has to be transmitted, since it would be multiplied by a zero on the receiving side anyway.
- the adaptive gain factor no longer needs to be transmitted.
- the fixed gain factor could be quantized, for example, scalar.
- adaptive codebook entry basic speech frequency
- adaptive gain factor can even be omitted in the case of an onset.
- each of these possible implementations is that a smaller number of bits can be transmitted compared to the state-of-the-art.
- these bits can now be used to improve the quantization of the fixed gain factor and / or the quantization of the fixed excitation and / or the quantization of the LPC coefficients.
- any remaining codec parameter can potentially benefit from improved quantization.
- no new parameter is provided (ie no second fixed code book), but instead the improved quantization of existing parameters. This saves computational complexity, memory requirements and enables the consideration of specific characteristics of subframes with onsets.
- By cleverly embedding the additional usable bits in the quantization tables of other codec parameters coding can also be memory-efficient.
- a skilful embedding of the additional bits that are released will be briefly outlined below. Assume that the adaptive excitation is set to zero by a reserved word in the adaptive code book. Then the fixed gain factor, which previously had 7 bits together with 'the ad- aptive gain factor was vector-quantized, with approximately the same quantization error, for example scalarized with 5 bits. The values of the fixed gain factor quantized with 5 bits could result from a 25% subset of the 7 bit vector codebook, namely a subset that can be addressed with any 5 bits from the 7 bits. Such a realization of the 5-bit scalar quantizer saves additional memory. The released 2 bits can now be used, for example, for more precise quantization of the fixed excitation.
Landscapes
- Engineering & Computer Science (AREA)
- Computational Linguistics (AREA)
- Signal Processing (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
Description
Claims
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE10124420A DE10124420C1 (en) | 2001-05-18 | 2001-05-18 | Coding method for transmission of speech signals uses analysis-through-synthesis method with adaption of amplification factor for excitation signal generator |
DE10124420 | 2001-05-18 | ||
PCT/DE2002/001598 WO2002095734A2 (en) | 2001-05-18 | 2002-05-02 | Method for controlling the amplification factor of a predictive voice encoder |
Publications (2)
Publication Number | Publication Date |
---|---|
EP1388146A2 true EP1388146A2 (en) | 2004-02-11 |
EP1388146B1 EP1388146B1 (en) | 2007-11-28 |
Family
ID=7685379
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP02740316A Expired - Fee Related EP1388146B1 (en) | 2001-05-18 | 2002-05-02 | Method for encoding and transmitting voice signals |
Country Status (5)
Country | Link |
---|---|
US (1) | US20040148162A1 (en) |
EP (1) | EP1388146B1 (en) |
CN (1) | CN100508027C (en) |
DE (2) | DE10124420C1 (en) |
WO (1) | WO2002095734A2 (en) |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US8515744B2 (en) | 2008-12-31 | 2013-08-20 | Huawei Technologies Co., Ltd. | Method for encoding signal, and method for decoding signal |
Families Citing this family (10)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
DE102005000828A1 (en) * | 2005-01-05 | 2006-07-13 | Siemens Ag | Method for coding an analog signal |
US7546237B2 (en) * | 2005-12-23 | 2009-06-09 | Qnx Software Systems (Wavemakers), Inc. | Bandwidth extension of narrowband speech |
BRPI0718300B1 (en) * | 2006-10-24 | 2018-08-14 | Voiceage Corporation | METHOD AND DEVICE FOR CODING TRANSITION TABLES IN SPEAKING SIGNS. |
CN101286319B (en) * | 2006-12-26 | 2013-05-01 | 华为技术有限公司 | Speech coding system to improve packet loss repairing quality |
US8688437B2 (en) | 2006-12-26 | 2014-04-01 | Huawei Technologies Co., Ltd. | Packet loss concealment for speech coding |
US8515767B2 (en) * | 2007-11-04 | 2013-08-20 | Qualcomm Incorporated | Technique for encoding/decoding of codebook indices for quantized MDCT spectrum in scalable speech and audio codecs |
KR101701081B1 (en) * | 2013-01-29 | 2017-01-31 | 프라운호퍼-게젤샤프트 츄어 푀르더룽 데어 안게반텐 포르슝에.파우. | Apparatus and method for selecting one of a first audio encoding algorithm and a second audio encoding algorithm |
CA2927716C (en) | 2013-10-18 | 2020-09-01 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Concept for encoding an audio signal and decoding an audio signal using speech related spectral shaping information |
BR112016008544B1 (en) * | 2013-10-18 | 2021-12-21 | Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. | ENCODER TO ENCODE AND DECODER TO DECODE AN AUDIO SIGNAL, METHOD TO ENCODE AND METHOD TO DECODE AN AUDIO SIGNAL. |
FR3013496A1 (en) * | 2013-11-15 | 2015-05-22 | Orange | TRANSITION FROM TRANSFORMED CODING / DECODING TO PREDICTIVE CODING / DECODING |
Family Cites Families (11)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5657418A (en) * | 1991-09-05 | 1997-08-12 | Motorola, Inc. | Provision of speech coder gain information using multiple coding modes |
SE504397C2 (en) * | 1995-05-03 | 1997-01-27 | Ericsson Telefon Ab L M | Method for amplification quantization in linear predictive speech coding with codebook excitation |
GB2312360B (en) * | 1996-04-12 | 2001-01-24 | Olympus Optical Co | Voice signal coding apparatus |
US6104992A (en) * | 1998-08-24 | 2000-08-15 | Conexant Systems, Inc. | Adaptive gain reduction to produce fixed codebook target signal |
US6330531B1 (en) * | 1998-08-24 | 2001-12-11 | Conexant Systems, Inc. | Comb codebook structure |
US6192335B1 (en) * | 1998-09-01 | 2001-02-20 | Telefonaktieboiaget Lm Ericsson (Publ) | Adaptive combining of multi-mode coding for voiced speech and noise-like signals |
US6691092B1 (en) * | 1999-04-05 | 2004-02-10 | Hughes Electronics Corporation | Voicing measure as an estimate of signal periodicity for a frequency domain interpolative speech codec system |
EP1088304A1 (en) * | 1999-04-05 | 2001-04-04 | Hughes Electronics Corporation | A frequency domain interpolative speech codec system |
US6782360B1 (en) * | 1999-09-22 | 2004-08-24 | Mindspeed Technologies, Inc. | Gain quantization for a CELP speech coder |
US6574593B1 (en) * | 1999-09-22 | 2003-06-03 | Conexant Systems, Inc. | Codebook tables for encoding and decoding |
US6510407B1 (en) * | 1999-10-19 | 2003-01-21 | Atmel Corporation | Method and apparatus for variable rate coding of speech |
-
2001
- 2001-05-18 DE DE10124420A patent/DE10124420C1/en not_active Expired - Fee Related
-
2002
- 2002-05-02 EP EP02740316A patent/EP1388146B1/en not_active Expired - Fee Related
- 2002-05-02 WO PCT/DE2002/001598 patent/WO2002095734A2/en active IP Right Grant
- 2002-05-02 CN CN02814429.5A patent/CN100508027C/en not_active Expired - Fee Related
- 2002-05-02 DE DE50211294T patent/DE50211294D1/en not_active Expired - Lifetime
- 2002-05-02 US US10/478,142 patent/US20040148162A1/en not_active Abandoned
Non-Patent Citations (1)
Title |
---|
See references of WO02095734A3 * |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US8515744B2 (en) | 2008-12-31 | 2013-08-20 | Huawei Technologies Co., Ltd. | Method for encoding signal, and method for decoding signal |
Also Published As
Publication number | Publication date |
---|---|
EP1388146B1 (en) | 2007-11-28 |
CN1533564A (en) | 2004-09-29 |
DE10124420C1 (en) | 2002-11-28 |
US20040148162A1 (en) | 2004-07-29 |
CN100508027C (en) | 2009-07-01 |
WO2002095734A2 (en) | 2002-11-28 |
WO2002095734A3 (en) | 2003-11-20 |
DE50211294D1 (en) | 2008-01-10 |
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