EP1989705A2 - Method for limiting adaptive excitation gain in an audio decoder - Google Patents
Method for limiting adaptive excitation gain in an audio decoderInfo
- Publication number
- EP1989705A2 EP1989705A2 EP07731604A EP07731604A EP1989705A2 EP 1989705 A2 EP1989705 A2 EP 1989705A2 EP 07731604 A EP07731604 A EP 07731604A EP 07731604 A EP07731604 A EP 07731604A EP 1989705 A2 EP1989705 A2 EP 1989705A2
- Authority
- EP
- European Patent Office
- Prior art keywords
- gain
- adaptive excitation
- error indication
- error
- decoder
- 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
- 230000005284 excitation Effects 0.000 title claims abstract description 86
- 230000003044 adaptive effect Effects 0.000 title claims abstract description 54
- 238000000034 method Methods 0.000 title claims description 46
- 230000007774 longterm Effects 0.000 claims abstract description 40
- 230000005540 biological transmission Effects 0.000 claims abstract description 21
- 230000005236 sound signal Effects 0.000 claims abstract description 8
- 238000012886 linear function Methods 0.000 claims description 3
- 230000006870 function Effects 0.000 description 14
- 238000001914 filtration Methods 0.000 description 10
- 239000013598 vector Substances 0.000 description 8
- 230000015572 biosynthetic process Effects 0.000 description 7
- 238000003786 synthesis reaction Methods 0.000 description 7
- 230000015556 catabolic process Effects 0.000 description 4
- 230000001186 cumulative effect Effects 0.000 description 4
- 238000006731 degradation reaction Methods 0.000 description 4
- 238000010586 diagram Methods 0.000 description 4
- 230000000694 effects Effects 0.000 description 3
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- 101100379142 Mus musculus Anxa1 gene Proteins 0.000 description 1
- 238000009825 accumulation Methods 0.000 description 1
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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
-
- 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
- G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
-
- 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
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
Definitions
- the present invention relates to a method of limiting adaptive excitation gain in a decoder of an audio signal. It also relates to a decoder of an audio signal encoded by means of an encoder comprising a long-term predictive filter.
- the invention finds an advantageous application in the field of coding and decoding of digital signals such as audio-frequency signals.
- the invention is particularly well suited to the transmission of speech and / or audio signals over packet networks, of the VoIP type, for example, to provide an acceptable quality during decoding after a loss of packets, in particular avoiding the saturation of the data.
- An example of a CELP encoder is the G.729 system recommended in I 1 ITU-T, designed for voiceband speech between 300 and 3400 Hz sampled at 8 kHz and transmitted at a fixed rate of 8 kbit / s. with frames of 10 ms. The detailed operation of this coder is specified in the article by R. Salami, C. Laflamme, JP Adoul, A. Kataoka, S.
- Figure 1 shows a high level view of a G.729 encoder. This figure shows a preprocessing high-pass filtering 101 for eliminating the signals with a frequency lower than 50 Hz. The thus filtered speech signal S (n) is then analyzed by the block 102 in order to determine a filter. z) Linear Prediction Coding (LPC), which is transmitted to the multiplexer 104 as an index indexing the quantized vector (QV) in a dictionary.
- LPC Linear Prediction Coding
- the original signal S (n) filtered by the filter ⁇ (z), called excitation, is processed by block 103 so as to extract the parameters mentioned in the table of FIG. 2. These parameters are then coded and transmitted to MUX multiplexer 104.
- the operation of the excitation coding block 103 is detailed in FIG. 1 (b). As can be seen in this figure, the excitation is coded in three steps: in a first step, a long-term prediction filtering (LTP) is performed by the blocks 106, 107, 11.
- LTP long-term prediction filtering
- the LTP filter of the G.729 encoder is a filter of order equal to 1.
- the residual difference between these two signals is modeled, on the one hand, by a fixed code c (n), or innovative code, extracted from an innovatory dictionary ACELP 108 with four pulses ⁇ 1, and, on the other hand, by a gain g c of fixed excitation 109.
- the fixed code c (n) and the gain g c are determined by minimizing in 1 1 1 'the error between the residual signal resulting from the previous LTP stage and the signal g c .c (n),
- the resulting parameters namely the pitch period P, the fixed code c (n) and the gains g p and g c of pitch fixed excitation, are encoded and transmitted to multiplexer 104.
- FIG. 1 (c) shows how a conventional G.729 decoder reconstructs the speech signal from the data received from the multiplexer 104 by the demultiplexer 1 12.
- the excitation is reconstituted by 5 ms subframes by adding two contributions:
- the excitation thus decoded is shaped by the synthesis filter 120 LPC 1 / ⁇ (z) whose coefficients are decoded by the block 1 19 in the domain of spectral line pairs (LSF) and interpolated by subframe of 5 ms.
- the reconstructed signal is then processed by an adaptive post-filter 121 and a post-processing high-pass filter 122.
- the decoder of FIG. 1 (c) thus relies on the source-filter model to synthesize the signal.
- the CELP type encoders In the case of excitation from the long-term prediction LTP filter, and in order to generate an excitation signal capable of rapidly following the signal attacks, the CELP type encoders generally allow the choice of a gain g p pitch greater than 1. As a result, the decoder is locally unstable. However, this instability is controlled by the synthesis analysis model which permanently minimizes the difference between the LTP excitation signal and the original target signal. During transmission or frame loss errors, this instability can lead to significant degradation due to the offset between the encoder and the decoder.
- the gain value g p of pitch not received in a frame is generally replaced by the value of g p in the previous frame, and although the variable nature of the speech signal consists of an alternation periods of voices with a pitch gain close to 1 and unvoiced with a pitch gain of less than 1 allows, in general, to limit the potential problems related to this local instability, it remains nonetheless true that, for some signals, including the voiced signals, transmission errors in periodic stationary areas can cause significant impairments when for example the replacement gain g p is higher than the actual gain and the affected frame is followed by high gain frames , as happens during attacks. This situation can then quickly cause a saturation of the LTP filter by cumulative effect related to the recursive character of long-term predictive filtering.
- a first solution to this problem is to limit the gp pitch gain to 1, but this constraint has the effect of degrading the performance of the CELP coders for attacks.
- the solution implemented in the G.723.1 standard is to find, by learning, for each vector of possible gain of the encoder an equivalent average gain of order 1. These values are stored in a table. This equivalent filter of order 1 is then used to estimate the maximum potential error accumulated in the long-term filter, and thus to identify the unstable zones where the gain must be limited in the event of a large accumulated error and to calculate the gain at apply to make the filter stable.
- the technical problem to be solved by the object of the present invention is to propose a method for limiting adaptive excitation gain in a decoder of an audio signal encoded by means of an encoder comprising a long-term predictive filter. , following a loss of transmission frame between said encoder and said decoder, which would limit the adaptive excitation g p gain, or pitch gain, only in the case where an instability LTP filter is actually noted, and to ensure the best possible compromise between the quality of the decoding and its robustness vis-à-vis the frame loss.
- frame loss is used here to mean non-reception of a frame as well as transmission errors in a frame.
- said arbitrary value is equal to a value of the adaptive excitation gain determined during said lost frame by an error concealment algorithm.
- said arbitrary value is equal to the value of the adaptive excitation gain for the non-lost frame preceding said lost frame.
- said arbitrary value is defined from a voicing detection of the previous frame.
- said arbitrary value is equal to 1, otherwise the arbitrary value is equal to 0.
- the excitation is composed of a random noise.
- the method according to the invention has the advantage of modifying the gain g p of pitch only when a possible instability of the LTP filter is detected at the decoder itself and not at the encoder as in known techniques.
- the method of the invention takes into account both the actual state of the decoder and the exact information on the transmission errors reached.
- the method, object of the invention can be used autonomously, that is to say in coding structures that do not provide for limiting the pitch gain at the coder.
- the invention provides that said adaptive excitation gain is supplied to said decoder by an encoder equipped with a gain limitation device.
- the method according to the invention can therefore also be used in combination with a known taming technique, installed at the encoder.
- the advantages of the two techniques are then cumulated: the prior art technique makes it possible to limit the too long sequences of pitch gains greater than 1. Indeed, such sequences cause a large propagation of the error, constraining the method of the invention. to modify the signal over long periods.
- a threshold too low tripping of the technique of "taming" a priori degrades the signal.
- the invention thus makes it possible to reduce the number of triggers of the "taming" technique a priori by increasing the threshold, because even if this technique does not detect the risk of explosion, the posterior method according to the invention detects it and cure it.
- said error indication function is of the form:
- x t (n) e t (n) + ⁇ g it .x t (n-P + i) i € [- (Nl) / 2, (Nl) / 2] i
- - iV is the order of the long-term, generally odd, predictive filter
- the gains g it are equal to the gains of adaptive excitation g t of said long-term predictive filter for the received frames or to the adaptive excitation gains g LFE c (FEC for "Frame Erasure Concealment") of said long-term predictive filter in the previous frame for lost frames,
- P is the period of adaptive excitation.
- the order iV of the LTP filter can be taken as 1.
- the gain g p of adaptive excitation of a long-term predictive filter of order 1 is limited to the value 1 if said parameter of indication of error is greater than said given threshold.
- the invention provides that a correction factor is applied to the gains g. adaptive excitation of a long-term predictive filter of order greater than 1 if said error indication parameter is greater than said given threshold.
- said at least one adaptive excitation gain is limited by a linear function of said given threshold if said error indication parameter is greater than said threshold.
- a discriminator able to determine, as a function of the result provided by the comparator, a value of at least one adaptive excitation gain to be used by the decoder.
- Figure 1 (a) is a high-level diagram of a G.729 encoder.
- Fig. 1 (b) is a detailed diagram of the coding block of the encoder excitation of Fig. 1 (a).
- Fig. 1 (c) is a diagram of the decoder associated with the encoder of Fig. 1 (a).
- FIG. 2 is a table giving the various coding parameters of the coder of FIG. 1 (a).
- FIG. 3 is a diagram of a decoder according to the invention.
- excitation signal x e (n) from the excitation coding block 103 of FIG. 1 (a) and explained in FIG. 1 (b) is the sum of the adaptive excitation g p . x e (nP) and fixed excitation g c .c (n):
- - g p is the gain of the adaptive excitation or pitch gain
- - P is the value of the pitch or length of the period.
- the G.729 encoder uses fractional resolution in 1/3 increments for small pitch values (P ⁇ 85) to better model high-pitched voices. Adaptive excitation with a fractional pitch is obtained by interpolation with oversampling,
- - g c is the gain of the fixed excitation
- - c (n) is the fixed code word, or innovator.
- Adaptive excitation depends solely on the past excitation and makes it possible to efficiently model the periodic signals, especially voiced signals, where the excitation itself is repeated almost periodically.
- the fixed part c (n) brings the innovation in the total excitation to model the difference between the periods, that is to say to correct the error between the adaptive excitation and the prediction residue.
- this excitation signal is optimized to the encoder using the technique of synthesis analysis.
- the synthesis filtering of this excitation is thus performed with the quantized filter to check the result that will be obtained at the decoder.
- the error concealment algorithm uses an estimated excitation signal from the past excitation signal.
- the decoder comprises a processing line of the excitation signal coming from the demultiplexer 1 12 constituted by the blocks 21 1 to 215.
- This decoder processing line thus described also serves to illustrate the main steps of the method of limiting the adaptive excitation gain according to the invention.
- the block 21 1 is intended to detect whether a frame is correctly received or not.
- This detection block is followed by a module 212 which performs a similar operation to LTP long-term filtering. More precisely, the module 212 calculates an error indication function x t (n) whose values are representative of the accumulated error on decoding on the adaptive excitation as a result of a loss of transmission. In one embodiment, this function is given by:
- xt (n) gt-x t (np) + e t (n)
- g t is equal to:
- a module 213 calculates from the values of the function x t ( ⁇ ) provided by the module 212, an error indication parameter S t .
- a comparator 214 checks whether the parameter S 1 does not exceed a certain threshold S 0 . In case of overshoot and if the decoded pitch gain g p is greater than 1, the value of g p is limited, because in this case there is a risk of saturation of the LTP filter.
- the error indication parameter S t may be the sum of the values of the function x t (n), or the maximum value, the average or the sum of the squares of these values.
- the comparator 214 is followed by a discriminator 215 able to determine the value g ' t of the pitch gain to be applied to the block 1 17 for the current frame, namely the decoded pitch value ⁇ or a limited value.
- the gain g 1 can be systematically limited to 1 for example, regardless of the magnitude of the overshoot. But we can also provide a more progressive limitation which consists in defining the gain g ' t as a linear function of the parameter S 1 of the form:
- the LTP, P and g p parameters are transmitted for each 5 ms subframe containing 40 samples.
- the treatment to avoid saturation of the LTP filter, which is the subject of the invention, is also carried out at the rate of the subframes.
- the parameter S t of error indication for example the sum of the function x t (n), is calculated for each subframe.
- the value of this parameter is limited to 120, which corresponds to an average value of 3: 39
- the memory of the signal x t ( ⁇ ) is updated with the new value g ⁇ .
- the long-term filter of the encoder is a filter of order 1.
- the pseudo-LTP filter used to define the error indication function may be the equivalent filter of order 1 or more advantageously a filter identical to that used in the encoder, in particular of the same order.
- the gain g ' t can be calculated in the same way as for a filter of order 1. It then applies the corrective factor g ' t / g e at the gains g, - of the higher order filter.
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- 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 (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR0650688A FR2897977A1 (en) | 2006-02-28 | 2006-02-28 | Coded digital audio signal decoder`s e.g. G.729 decoder, adaptive excitation gain limiting method for e.g. voice over Internet protocol network, involves applying limitation to excitation gain if excitation gain is greater than given value |
PCT/FR2007/050779 WO2007099244A2 (en) | 2006-02-28 | 2007-02-13 | Method for limiting adaptive excitation gain in an audio decoder |
Publications (2)
Publication Number | Publication Date |
---|---|
EP1989705A2 true EP1989705A2 (en) | 2008-11-12 |
EP1989705B1 EP1989705B1 (en) | 2012-08-15 |
Family
ID=36407997
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
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EP07731604A Not-in-force EP1989705B1 (en) | 2006-02-28 | 2007-02-13 | Method for limiting adaptive excitation gain in an audio decoder |
Country Status (7)
Country | Link |
---|---|
US (1) | US8180632B2 (en) |
EP (1) | EP1989705B1 (en) |
JP (1) | JP4988774B2 (en) |
KR (1) | KR101372460B1 (en) |
CN (1) | CN101395659B (en) |
FR (1) | FR2897977A1 (en) |
WO (1) | WO2007099244A2 (en) |
Families Citing this family (17)
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US7877253B2 (en) * | 2006-10-06 | 2011-01-25 | Qualcomm Incorporated | Systems, methods, and apparatus for frame erasure recovery |
CN101604525B (en) * | 2008-12-31 | 2011-04-06 | 华为技术有限公司 | Pitch gain obtaining method, pitch gain obtaining device, coder and decoder |
WO2011047887A1 (en) * | 2009-10-21 | 2011-04-28 | Dolby International Ab | Oversampling in a combined transposer filter bank |
CN101969372B (en) * | 2010-10-29 | 2012-11-28 | 上海交通大学 | Frame loss prediction based cellular network uplink video communication QoS (Quality of Service) optimization method |
PT2676267T (en) | 2011-02-14 | 2017-09-26 | Fraunhofer Ges Forschung | Encoding and decoding of pulse positions of tracks of an audio signal |
MY164797A (en) | 2011-02-14 | 2018-01-30 | Fraunhofer Ges Zur Foederung Der Angewandten Forschung E V | Apparatus and method for processing a decoded audio signal in a spectral domain |
CN103620672B (en) * | 2011-02-14 | 2016-04-27 | 弗劳恩霍夫应用研究促进协会 | For the apparatus and method of the error concealing in low delay associating voice and audio coding (USAC) |
BR112012029132B1 (en) | 2011-02-14 | 2021-10-05 | Fraunhofer - Gesellschaft Zur Förderung Der Angewandten Forschung E.V | REPRESENTATION OF INFORMATION SIGNAL USING OVERLAY TRANSFORMED |
AU2012217216B2 (en) | 2011-02-14 | 2015-09-17 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for coding a portion of an audio signal using a transient detection and a quality result |
EP2676266B1 (en) | 2011-02-14 | 2015-03-11 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Linear prediction based coding scheme using spectral domain noise shaping |
KR102138320B1 (en) | 2011-10-28 | 2020-08-11 | 한국전자통신연구원 | Apparatus and method for codec signal in a communication system |
US9449607B2 (en) | 2012-01-06 | 2016-09-20 | Qualcomm Incorporated | Systems and methods for detecting overflow |
US9842598B2 (en) | 2013-02-21 | 2017-12-12 | Qualcomm Incorporated | Systems and methods for mitigating potential frame instability |
EP2922055A1 (en) | 2014-03-19 | 2015-09-23 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus, method and corresponding computer program for generating an error concealment signal using individual replacement LPC representations for individual codebook information |
EP2922054A1 (en) | 2014-03-19 | 2015-09-23 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus, method and corresponding computer program for generating an error concealment signal using an adaptive noise estimation |
EP2922056A1 (en) | 2014-03-19 | 2015-09-23 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus, method and corresponding computer program for generating an error concealment signal using power compensation |
EP2980795A1 (en) * | 2014-07-28 | 2016-02-03 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Audio encoding and decoding using a frequency domain processor, a time domain processor and a cross processor for initialization of the time domain processor |
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US5479559A (en) * | 1993-05-28 | 1995-12-26 | Motorola, Inc. | Excitation synchronous time encoding vocoder and method |
US5708757A (en) * | 1996-04-22 | 1998-01-13 | France Telecom | Method of determining parameters of a pitch synthesis filter in a speech coder, and speech coder implementing such method |
US5960386A (en) * | 1996-05-17 | 1999-09-28 | Janiszewski; Thomas John | Method for adaptively controlling the pitch gain of a vocoder's adaptive codebook |
US5987406A (en) * | 1997-04-07 | 1999-11-16 | Universite De Sherbrooke | Instability eradication for analysis-by-synthesis speech codecs |
JP4464488B2 (en) * | 1999-06-30 | 2010-05-19 | パナソニック株式会社 | Speech decoding apparatus, code error compensation method, speech decoding method |
US6636829B1 (en) * | 1999-09-22 | 2003-10-21 | Mindspeed Technologies, Inc. | Speech communication system and method for handling lost frames |
US6574593B1 (en) * | 1999-09-22 | 2003-06-03 | Conexant Systems, Inc. | Codebook tables for encoding and decoding |
CA2388439A1 (en) * | 2002-05-31 | 2003-11-30 | Voiceage Corporation | A method and device for efficient frame erasure concealment in linear predictive based speech codecs |
JP4733939B2 (en) * | 2004-01-08 | 2011-07-27 | パナソニック株式会社 | Signal decoding apparatus and signal decoding method |
WO2006009074A1 (en) * | 2004-07-20 | 2006-01-26 | Matsushita Electric Industrial Co., Ltd. | Audio decoding device and compensation frame generation method |
WO2006098274A1 (en) * | 2005-03-14 | 2006-09-21 | Matsushita Electric Industrial Co., Ltd. | Scalable decoder and scalable decoding method |
US7831421B2 (en) * | 2005-05-31 | 2010-11-09 | Microsoft Corporation | Robust decoder |
CN101213590B (en) * | 2005-06-29 | 2011-09-21 | 松下电器产业株式会社 | Scalable decoder and disappeared data interpolating method |
-
2006
- 2006-02-28 FR FR0650688A patent/FR2897977A1/en not_active Withdrawn
-
2007
- 2007-02-13 US US12/224,566 patent/US8180632B2/en not_active Expired - Fee Related
- 2007-02-13 KR KR1020087023810A patent/KR101372460B1/en active IP Right Grant
- 2007-02-13 EP EP07731604A patent/EP1989705B1/en not_active Not-in-force
- 2007-02-13 CN CN2007800071077A patent/CN101395659B/en not_active Expired - Fee Related
- 2007-02-13 WO PCT/FR2007/050779 patent/WO2007099244A2/en active Application Filing
- 2007-02-13 JP JP2008556824A patent/JP4988774B2/en not_active Expired - Fee Related
Non-Patent Citations (1)
Title |
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See references of WO2007099244A2 * |
Also Published As
Publication number | Publication date |
---|---|
JP4988774B2 (en) | 2012-08-01 |
KR101372460B1 (en) | 2014-03-11 |
JP2009528563A (en) | 2009-08-06 |
CN101395659A (en) | 2009-03-25 |
EP1989705B1 (en) | 2012-08-15 |
KR20080102262A (en) | 2008-11-24 |
WO2007099244A2 (en) | 2007-09-07 |
CN101395659B (en) | 2012-11-07 |
FR2897977A1 (en) | 2007-08-31 |
WO2007099244A3 (en) | 2007-10-25 |
US20090204412A1 (en) | 2009-08-13 |
US8180632B2 (en) | 2012-05-15 |
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