Patent Application
20080126081
The invention relates to a method as well as a device for the artificial extension of the bandwidth of speech signals.
Speech signals cover a wide frequency range that extends from the fundamental speech frequency, which depending on the speaker lies in the range between 80 to 160 Hz, up to the frequencies beyond 10 kHz. However, during speech communication via particular transmission media, such as telephones for example, only a limited segment is transmitted for reasons of bandwidth efficiency, whereby a sentence intelligibility of approximately 98% is ensured.
Corresponding to the minimum bandwidth from 300 Hz to 3.4 kHz specified for the telephone system, a speech signal can essentially be divided into three frequency ranges. In this way, each of these frequency ranges characterizes specific speech properties as well as subjective perceptions. Thus lower frequencies below approximately 300 Hz primarily arise during sonorous speech segments such as vowels, for example. In this case, this frequency range contains tonal components, which in particular means the fundamental speech frequency as well as several possible harmonics, depending on the pitch of the voice.
These low frequencies are important for the subjective perception of the volume and dynamics of a speech signal. In contrast, the fundamental speech frequency can be perceived by a human listener as a result of the psycho-acoustic property of virtual pitch perception from the harmonic structure in higher frequency ranges even if the low frequencies are missing. Thus medium frequencies in the range from approximately 300 Hz to approximately 3.4 kHz are basically present in the speech signal during speech activities. Their time-variant spectral coloration by multiple formants as well as the temporal and spectral fine structure characterizes the spoken sound or phoneme in each instance. In such a manner, the medium frequencies transport the main part of the information relevant for the intelligibility of the speech.
Alternatively, high frequency rates above approximately 3.4 kHz develop during unvoiced sounds, as is particularly strongly the case during sharp sounds such as “s” or “f”, for example. In addition, so-called plosive sounds like “k” or “t” have a wide spectrum with strong high-frequency rates. Therefore, the signal has more of a noisy character than a tonal character in this upper frequency range. The structure of the formants that are also present in this range is relatively time-invariant, but varies for different speakers. The high frequency rates are of considerable importance for clarity, presence and naturalness of a speech signal, because without these high frequency rates the speech sounds dull. Furthermore, superior differentiation between fricatives and consonants is made possible by high frequency rates of this type, whereby these high frequency rates also thereby ensure increased intelligibility of the speech.
During a transmission of a speech signal via a speech communications system comprising a transmission channel with a limited bandwidth, in principle it is desired and is always the goal that the speech signal to be transmitted be capable of transmission with the best-possible quality from a transmitter to a receiver. Here the speech quality is however a subjective variable with a plurality of components, of which the intelligibility of the speech signal represents the most important for a speech communications systems of this type.
A relatively high level of speech intelligibility can already be achieved with modern digital transmission systems. At the same time, it is known that an improvement in the subjective assessment of the speech signal is made possible by an extension of the telephone bandwidth at high frequencies (higher than 3.4 kHz) as well as at low frequencies (lower than 300 Hz). In terms of a subjective quality improvement, a bandwidth increased in comparison to the normal telephone bandwidth is to be targeted for systems for speech communication. One possible approach relates to in modifying the transmission and in effecting a wider transmitted bandwidth by an encoding method, or alternatively in performing an artificial bandwidth extension. Through an extension of the bandwidth of this type, the frequency bandwidth on the receiver side is widened to the range from 50 Hz to 7 kHz. Suitable signal processing algorithms allow parameters to be determined for the wideband model from short segments of a narrowband speech signal using methods of pattern recognition, said parameters then being used to estimate the missing signal components for the speech. With the method, a wideband equivalent with frequency components in the range 50 Hz to 7 kHz is created from the narrowband speech signal, and an improvement in the subjectively perceived speech quality is effected.
In current speech signal and audio signal encoding algorithms, additional techniques of artificial bandwidth extension are used. For example, in the wideband range (acoustic bandwidth of 50 Hz to 7 kHz) speech encoding standards such as the AMR-WB (Adaptive Multirate Wideband) encoding-decoding algorithm are used. With this AMR-WB standard, upper frequency subbands (frequency range of approximately 6.4 to 7 kHz) are extrapolated from lower frequency components. In encoding-decoding methods of this type, the bandwidth extension is generally produced by a comparatively small amount of ancillary information. This ancillary information can be filter coefficients or amplification factors for instance, whereby the filter coefficients can be produced by an LPC (Linear Prediction Filter) method for example. This ancillary information is transmitted to a receiver in an encoded bitstream. Other standards which are based on the extension of the bandwidth technique can currently be seen in the standards AMR-WB+ and the extended aacPlus speech/audio encoding-decoding method. Methods that are designed to encode and decode information are called codecs and include both an encoder as well as a decoder. Every digital telephone, regardless of whether it is designed for a fixed network or a mobile radio network, contains a codec of the type that converts analogue signals into digital signals, and digital signals into analogue signals. A codec of this type can be implemented in hardware or in software.
In current implementations of speech/audio signal encoding algorithms in which the technology for bandwidth extension is used, components of an extension band, for example in the frequency range from 6.4 to 7 kHz, are encoded and decoded by the LPC encoding technology already mentioned. In doing so, an LPC analysis of the extension band of the input signal is carried out in an encoder, and the LPC coefficients as well as the amplification factors are encoded from subframes of a residual signal. The residual signal of the extension band is produced in a decoder, and the transmitted amplification factors and the LPC synthesis filters are used for the generation of an output signal. The approach described above can be used either directly on the wideband input signal or even with a subband signal from the extension band downsampled at a threshold or in a critical range.
In the extended aacPlus encoding standard, the SBR (Spectral Band Replication) technique is used. At the same time, the wideband audio signal is split into frequency subbands by a 64-channel QMF filter bank. For the high-frequency filter bank channels, a sophisticated and technically highly developed parametric encoding is applied to the subbands of the signal components, whereby a large number of detectors and estimators are necessary for this purpose, which are used in order to control the bitstream content. Even though an improvement, in particular in the speech quality of speech signals, can already be achieved using the known standards and encoding-decoding methods, an additional improvement in this speech quality is nevertheless to be targeted. Furthermore, the standards and encoding-decoding methods described above are very time-consuming and have a very complex structure.
As such, the one possible object of the present invention is to provide a method and a device for the artificial extension of the bandwidth of speech signal, with which improved speech quality and improved speech intelligibility can be achieved. Furthermore, this should be able to be implemented in a relatively simple and inexpensive manner.
The following steps are carried out in a method proposed by the inventors, for the artificial extension of the bandwidth of speech signals:
The method allows an improvement in the speech intelligibility and the speech quality during the transmission of speech signals to be achieved, with audio signals also being considered as speech signals. Furthermore, the method is also very robust with respect to disruptions during transmission.
The signal components necessary for bandwidth extension are advantageously determined from the wideband input speech signal by filtering, in particular bandpass filtering, whereby a simple and inexpensive selection of the necessary signal components can be carried out.
The determination of the temporal envelopes in step c) is preferably carried out independently of the determination of the spectral envelopes in step d). The envelopes can thus be determined in a precise manner, whereby a mutual interaction can be avoided.
A quantization of the temporal envelopes and the spectral envelopes is preferably carried out prior to the encoding of the temporal envelopes and the spectral envelopes in step e). The signal powers are determined from spectral subbands of the signal components determined for the bandwidth extension in an advantageous manner in step d) for the determination of the spectral envelopes. In this way, the temporal and spectral envelopes for the characterization can be determined very precisely.
In order to determine the signal powers of the spectral subbands, signal segments of the signal components determined for the bandwidth extension are generated in a preferred manner, with these signal segments in particular being transformed, in particular FF (Fast Fourier) transformed. In addition, the signal powers are determined from temporal signal segments of the signal components determined for the bandwidth extension in an advantageous manner in step c) for the determination of the temporal envelopes. The necessary parameters can herewith be determined in an inexpensive manner.
The encoded information relating to the forms to be reconstructed of the temporal envelopes and of the spectral envelopes are decoded in step f) in an advantageous manner.
An excitation signal is advantageously produced in a decoder from a signal transmitted to a decoder, with the transmitted signal comprising a signal power of this type in the frequency range that corresponds to that of the extension signal of the wideband input speech signal, which enables the production of an excitation signal. A modulated narrowband signal with a bandwidth with frequencies below the frequencies of the bandwidth of the extension band of the wideband input speech signal is preferably transmitted to the decoder for the production of the excitation signal. The excitation signal preferably has harmonics of the fundamental frequency of the signal transmitted to the decoder.
A first correction factor is advantageously determined from the decoded information of the temporal envelopes and the excitation signal. Furthermore, a reconstructed formation of the temporal envelopes is carried out from the first correction factor and the excitation signal, in particular by multiplying the first correction factor with the excitation signal. Furthermore, the reconstructed formation of the temporal envelopes is advantageously filtered, and pulse responses are produced at the time of filtering. A reconstructed formation of the spectral envelopes is carried out from the pulse responses and the reconstructed formation of the temporal envelopes. In addition, the signal components of the extension band of the wideband input speech signal are reconstructed from the reconstructed formation of the spectral envelopes. The reconstruction of the temporal and the spectral envelopes can herewith be carried out very reliably and very accurately.
A narrowband signal with a bandwidth with frequencies below the frequencies of the extension band of the wideband input signal is transmitted to the decoder in an advantageous embodiment.
The bandwidth-extended output speech signal is determined in an advantageous manner from the narrowband signal transmitted to the decoder and the reconstructed formation of the spectral envelopes, in particular from a summation of these two signals, and is provided as an output signal of the decoder. Thus an output signal can be created and provided, which ensures a high level of speech intelligibility and speech quality.
The steps a) through e) are preferably carried out in an encoder, which is preferably arranged in a transmitter. The encoded information produced in step e) is transmitted in an advantageous manner to the decoder as a digital signal. At least step f) is carried out in a preferred manner in a receiver, with the decoder being arranged in the receiver. However, it can also be provided that all steps a) through f) of the method are carried out in a receiver. In this case, the steps a) through e) are replaced in the receiver by an estimation process (to be implemented differently). The steps a) through e) can also be carried out separately in a transmitter.
The wideband input speech signal advantageously includes a bandwidth between approximately 50 Hz and approximately 7 kHz. The extension band of the wideband input speech signal preferably includes the frequency range of between approximately 3.4 kHz and approximately 7 kHz. In addition, the narrowband signal includes a signal range of the wideband input speech signal of approximately 50 Hz to approximately 3.4 kHz.
A device for the artificial extension of the bandwidth of speech signals, in which a wideband input speech signal can be placed, comprises at least the following components:
The device enables improved speech quality and improved speech intelligibility of speech signals during transmission in communications devices, such as mobile radio devices or ISDN devices for example.
The units a) through d) is advantageously embodied as an encoder. The encoder can be arranged in a transmitter or in a receiver, with the decoder being arranged in a receiver.
Advantageous embodiments of the method can also be considered advantageous embodiments of the device, where transferable.
These and other objects and advantages will become more apparent and more readily appreciated from the following description of the preferred embodiments, taken in conjunction with the accompanying drawings of which:
Reference will now be made in detail to the preferred embodiments, examples of which are illustrated in the accompanying drawings, wherein like reference numerals refer to like elements throughout.
The term ‘speech signals’ also includes audio signals as explained in greater detail below. In
Furthermore, it is also to be recognized from the illustration in
In addition, a block 2 is shown in
In the exemplary embodiment, the encoder 1 as well as blocks 2 and 3 are arranged in a first telephone device. The wideband input speech signal has a bandwidth of approximately 50 Hz to approximately 7 kHz in the exemplary embodiment. This wideband input speech signal swbi(k) is located in the bandpass filter or block 11 of the encoder 1, as can be inferred from the illustration in
This determination of the temporal envelopes as well as the spectral envelopes is explained in greater detail below. In this way, the signal seb(k) characterizing the signal components necessary for the bandwidth extension is first segmented, and this windowed signal segment is transformed. The segmentation of the signals seb(k) takes place in frames with a length of k sample values in each case. All subsequent steps and partial algorithms are carried out by frame consistently. Each speech frame (of 10 ms or 20 ms or 30 ms duration, for example) can be divided into multiple subframes (2.5 or 5 ms duration, for example) in an advantageous manner.
The windowed signal segments are then transformed. In the exemplary embodiment, a transformation is carried out here by a FFT (Fast Fourier Transform) in the frequency domain. The FFT transformed signal segments are determined here according to the following formula 1):
In this formula 1), Nf designates the FFT length or the frame size, μ designates the frame index and Mf designates the overlapping of the frames of the windowed signal segments. In addition, wf(κ) identifies the window function. The signal power in subbands of the frequency range of the extension band is then subsequently calculated in the frequency domain. This calculation of the signal strength or of the signal power is performed according to the following formula 2):
In this formula 2), λ designates the index of the corresponding subband, whereby EBλ characterizes the amount that contains all FFT interval ranges i with non-null coefficients in the λ frequency domain window wλ(i). The signal powers Pf(μ,λ) for the subbands according to formula 2) characterize the information of the spectral envelopes, which are transmitted to a decoder.
The determination of the temporal envelopes in the time domain is carried out in a manner similar to that for the determination of the spectral envelopes, and is based on short-term windowed segments of the bandpass-filtered wideband input speech signal swbi(k). Signal segments of the signal seb(k) are therefore taken into consideration during the determination of the temporal envelopes as well. The signal power is calculated for each windowed segment according to the following formula 3:
In this formula 3), Nt designates the frame length, v designates the frame index and Mt in turn designates the overlapping of the frames of the signal segments. It should be noted that, in general, the frame length Nt and the overlapping of the frames Mt, which are used for the extraction of the temporal envelopes, are smaller or much smaller than the corresponding figures Nf and Mf, which are used for the determination of the spectral envelopes.
An alternative for the extraction of the parameters of the temporal envelopes of the signal seb(k) can be seen in that a Hilbert transformation (90° phase shift filter) of the signal seb(k) is carried out. A summation of the short-segment signal powers of the filtered parts and of the original parts of the signal seb(k) results in the short-term temporal envelopes which are downsampled in order to determine the signal powers Pt(v). The signal powers Pt(v) of the signal segments then characterize the information for the temporal envelopes.
The signals sp
This digital signal BWE is transmitted to a decoder which is to be explained in greater detail below. It should be noted that a collective or associated encoding, as can be made possible by a vector quantization, for example, can be carried out in the case of a redundancy between the extracted parameters of the signal strengths according to formulas 2) and 3).
Furthermore, as can be seen from the illustration in
The signal components of a narrowband range of the wideband input speech signal swbi(k) are filtered by this block 2, which is embodied as a bandpass filter. The narrowband range lies between 50 Hz and 3.4 kHz in the exemplary embodiment. The output signal of block 2 is a narrowband signal snb(k) and is transmitted to block 3, which is embodied as an additional encoder in the exemplary embodiment. In this block 3, the narrowband signal snb(k) is encoded and transmitted as a bitstream to the decoder described below as a digital signal BWN.
In
Furthermore, the decoder 5, which is arranged in a receiver in the exemplary embodiment, has a block 52, which is designed for the decoding of the signal BWE transmitted between the encoder 1 and the decoder 2 via a transmission route. It is should be noted that even the digital signal BWN is transmitted via this transmission route between the encoder 1 and the decoder 5. As can be seen from the illustration in
As already addressed above, the information contained in the encoded digital signal BWE is decoded in block 52, and the signal powers that are calculated according to formulas 2) and 3), and which characterize the temporal envelopes and the spectral envelopes, are reconstructed. As can be seen from the illustration in
In the case of hierarchical speech encoding, there is an option of achieving this by using parameter of the additional decoder 4. For example, if Δk is a proportional or actual shift of the fundamental frequency and b of the LTB amplification factor for an adaptive code book in a CELP narrowband decoder, then an excitation with harmonic frequencies is possible, for example, during an integral multiplication of the momentary fundamental frequency through an LTP synthesis filtration by a bandpass filter (frequency range of the extension band) from an arbitrary signal neb(k).
At the same time, the FFT excitation signal emerges according to the following formula 4):
s
exc(k)=neb(k)+f(b)·sexc(k−Δk)
At the same time, the LTP amplification factor can be reduced or limited by the function f(b), in order to be able to prevent an overvoicing of the produced signal components of the extension band. It should be noted that a plurality of additional alternatives can be carried out in order to be able to carry out a synthetic wideband excitation by parameters of a narrowband codec.
An additional option for being able to produce an excitation signal relates to modulation of the narrowband signal snb(k) being carried out with a sine function at a fixed frequency, or through a direct use of an arbitrary signal neb(k), as was already defined above. It should be emphasized that the method that is used for the production of the excitation signal sexc(k) is completely independent of the generation of the digital signal BWE as well as the format of this digital signal BWE as well as the decoding of this digital signal BWE. As such, an independent adjustment can be carried out in this regard.
The reconstructed formation of the temporal envelopes is explained in greater detail below. As already addressed, the digital signal BWE is decoded in block 52, and the parameters characterizing the temporal envelopes and the spectral envelopes for the signal powers that are calculated according to formulas 2) and 3) are provided corresponding to the signals sp
As can be seen here from
In the exemplary embodiment shown according to
It should be noted that the embodiment shown in FIG is merely exemplary, and that even a single reconstructed formation of the temporal envelopes, as is carried out in the first decoder area 53, and a single reconstructed formation of the spectral envelopes, as is carried out in the second decoder area 54, is sufficient. It should likewise be noted that it can also be provided that the reconstructed formation of the spectral envelopes in the second decoder area 54 is carried out prior to the reconstructed formation of the temporal envelopes in the first decoder area 53. This means that in an embodiment of this type the second decoder area 54 is arranged upstream of the first decoder area 53. However, it can also be provided that the alternating performance of a reconstructed formation of the temporal envelopes and a reconstructed formation of the spectral envelopes is continued once more, and that an additional decoder area is subsequently arranged in the third decoder area 55 in the embodiment shown in
As already stated above, the proposed method and device are used in the exemplary embodiment in an advantageous manner for a wideband input speech signal with a frequency range of approximately 50 Hz to 7 kHz. Likewise, in the exemplary embodiment, the proposed method and device are provided for the artificial extension of the bandwidth of speech signals, whereby the extension band is determined by the frequency range of approximately 3.4 kHz to approximately 7 kHz when doing so. However, it can also be provided that the proposed method and device are used for an extension band that is located in a lower frequency range. In this way, the extension band can include a frequency range of approximately 50 Hz or even lower frequencies, up to a frequency range of approximately 3.4 kHz for example. It should be explicitly emphasized that the method for the artificial extension of the bandwidth of speech signals may also be used in such a manner that the extension band includes a frequency range that is above a frequency of approximately 7 kHz, at least in part, and up to 8 kHz for example, 10 kHz in particular, or even higher.
As already explained, a reconstructed formation for the temporal envelopes is generated in the first decoder area 53 according to
s′
exc(k)=g(k)·sexc(k);
S′
exc(z)=G(z)*Sexc(z)
As long as the spectral envelopes are not changed in principle by the first decoder area 53, the first scalar correction factor or amplification factor g1(k) has strict low-pass frequency characteristics.
For the calculation of these amplification factors or these first correction factors g1(k), the excitation signal sexc(k) is segmented and analyzed in the manner already carried out above for the segmentation and the analysis of the extraction of the temporal envelopes or the production of the signal Sp
The amplification factor or first correction factor g1(k) is calculated from this amplification factor γ(v) by interpolation and low-pass filtration. In this process, the low-pass filtration is of decisive importance for restricting the effect of this amplification factor or this first correction factor g1(k) to the spectral envelopes.
The reconstructed formation of the spectral envelopes of the necessary signal components of the extension band is determined by filtering the output signal s′exc(k), which characterizes the reconstructed formation of the temporal envelopes. At the same time, the filter operation can be implemented in the time domain or in the frequency domain. In order to be able to avoid a large time variation or time drift for the pulse response h(k), the corresponding frequency characteristic H(z) can be smoothed. In order to be able to determine the desired frequency characteristics, the output signal s′exc(k) of the first decoder area 53 is analyzed in order to be able to find the signal powers for Pfexc(μ, λ). The desired amplification factor Φ(μ, λ) of a corresponding subband of the frequency range of the extension band is calculated according to the following formula 7):
The frequency characteristic H(μ,i) of the form filter of the spectral envelopes can be calculated through an interpolation of the amplification factor Φ(μ,λ) and with a smoothing, taking the frequency into account. If the formation filter of the spectral envelopes are to be used in the time domain, for example through a linear-phase FIR filter, the filter coefficients can be calculated through an inverse FF transformation of the frequency characteristic H(λ,i) and a subsequent windowing.
As was explained and demonstrated in the examples above, the reconstructed formation of the temporal envelopes affects the reconstructed formation of the spectral envelopes and vice versa. It is therefore advantageous that, as explained in the exemplary embodiment and shown in
In the described exemplary embodiment according to
The encoder 1 as well as blocks 2 and 3 are advantageously arranged in a transmitter, whereby logically even the processes carried out in blocks 2 and 3 as well as the encoder 1 are then also carried out in the transmitter. Block 4 as well as decoder 5 can be advantageously arranged in this receiver, whereby it also clear that the previous steps carried out in decoder 5 and in block 4 are processed in the receiver. It should be noted that the proposed method and device can also be implemented in such a manner that the processes carried out in encoder 1 are carried out in decoder 5 and are thus exclusively carried out in the receiver. At the same time, it can be provided that the signal powers that are calculated according to formulas 2) and 3) are estimated in the decoder 5. At the same time, block 52 in particular is designed for the estimation of this parameter of the signal powers. This embodiment makes it possible to conceal potential transmission errors of the ancillary information transmitted in the digital signal BWE. Through a temporary estimation of lost parameters of an envelope, for example through data loss, an undesirable conversion of the signal bandwidth can be prevented.
Differing from the known methods for the artificial extension of the bandwidth of speech signals, with the proposed method no transmissions of already-used amplification factors and filter coefficients as ancillary information take place, but rather only the desired temporal and spectral envelopes are transmitted to a decoder as ancillary information. Amplification factors and filter coefficients are then first calculated in the decoder that is arranged in a receiver. The artificial extension of the bandwidth can be analyzed in this way in the receiver, and can be corrected, if necessary, in an inexpensive manner. Furthermore, the proposed method as well as the proposed device are very robust with respect to disruptions to the excitation signal, with a disruption of this type of a received narrowband signal being able to be generated by transmission errors.
Very good resolution or division can be achieved in the time domain and in the frequency domain by separately implementing the analysis, the transmission and the reconstructed shape of the temporal and spectral envelopes. Splitting in the time domain and the frequency domain may be achieved. This leads to very good reproducibility both of steady sounds and signals as well as of temporary or brief signals. For speech signals, the reproduction of stop consonants and plosives benefits from the significantly improved time resolution.
In contrast to known bandwidth extensions, the proposed method enables the frequency formation to be carried out by linear phase FIR filters instead of LPC synthesis filters. Typical artefacts (“filter ringing”) can also be reduced by doing so. Furthermore, the proposed method enables a very flexible and modular design, which furthermore makes it possible for the individual blocks in the receiver or in the decoder 5 to be exchanged or discontinued in a simple way. In an advantageous manner, no modification of the transmitter or the encoder 1 or of the format of the transmissions signal with which the encoded information is transmitted to the decoder 5 or the receiver is necessary for such a modification or discontinuation. Furthermore, different decoders may be operated with the proposed method, whereby a reproduction of the wideband input signal can be carried out with variable precision depending on the available computing power.
It should also be noted that the received parameters which characterize the spectral and temporal envelopes can be used not only for an extension of the bandwidth, but also for the support of subsequent signal processing blocks, such as a subsequent filtration, for example, or additional encoding steps such as transformation encoders can be used.
The resulting narrowband speech signal Snb(k), as is available to the algorithm for bandwidth extension, can exist after a reduction of the scanning frequency by a factor of 2 with a scanning rate of 8 kHz, for example.
With the proposed method and the underlying principle of bandwidth extension, it is possible to generate a wideband excitation of information for the G.729A+ standards. The data rates for the ancillary information transmitted in the digital signal BWE can amount to approximately 2 kbit/s. Furthermore, the proposed method requires a calculation system of relatively low complexity or a computational effort of relatively low complexity, which amounts to less than 3 WMOPS. Furthermore, the proposed method and the proposed device are very robust with respect to base-band disruptions of the G.729A+ standards. The principles can also be used in an advantageous manner for deployment in voice over IP. Furthermore, the method and the device are compatible with TDAC envelopes. Last but not least, the proposed method and device have a very modular and flexible design, and a modular and flexible concept.
A description has been provided with particular reference to preferred embodiments thereof and examples, but it will be understood that variations and modifications can be effected within the spirit and scope of the claims which may include the phrase “at least one of A, B and C” as an alternative expression that means one or more of A, B and C may be used, contrary to the holding in Superguide v. DIRECTV, 358 F3d 870, 69 USPQ2d 1865 (Fed. Cir. 2004).
| Number | Date | Country | Kind |
|---|---|---|---|
| 10 2005 032 724.9 | Jul 2005 | DE | national |
This application is based on and hereby claims priority to Application No. PCT/EP2006/063742 filed on Jun. 30, 2006 and DE Application No. 10 2005 032 724.9, filed on Jul. 13, 2005, the contents of which are hereby incorporated by reference.
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/EP06/63742 | 6/30/2006 | WO | 00 | 3/13/2007 |
