Method for calculating compensation coefficients for an analog reception chain, device and corresponding program

Abstract

Method for calculating compensation coefficients of an analog reception chain, device and corresponding program The present invention relates to a method for calculating static compensation coefficients of an analog reception chain, a method implemented by an electronic device during a calibration phase of a reception device comprising the analog reception chain and a digital compensation module implementing an FIR filter whose coefficients are the calculated static compensation coefficients, the method comprising: - an estimation step (A01), using a predetermined calibration signal, of a frequency response of the analog reception chain; - a selection step (A02), using the frequency response, of a bandwidth over which the static compensation coefficients are calculated;a calculation step (A03) of the static compensation coefficients of the analog reception chain, in the time domain, as a function of the bandwidth and the inverse of the frequency response over this bandwidth. Figure for the abbreviation: 3;

Description

Title of the invention: Method for calculating compensation coefficients of an analog reception chain, device and corresponding program [0001 ] 1. Field of the invention#

[0002] The invention relates to the reception of radiofrequency signals. The invention relates more particularly to the reception of radiofrequency signals within a reception device comprising an analog chain and a digital chain. An object of the invention is to perform compensation for distortions in an analog reception chain of a digital receiver comprising an analog reception chain and a digital reception chain. 1. Prior Art#

[0003] In many areas of communication, it is necessary to meet a requirement for increased communication speeds. This is the case, for example, in the civil field but also in the military field. This increase in speeds is made necessary by the ever-increasing use of communication devices to receive ever-increasing quantities of data.

[0004] The increase in data rates must be achieved while guaranteeing constant reception qualities. Thus, the continuous increase in data rates of military communications requires the use of waveforms that are increasingly demanding in terms of managing radio distortions, more particularly in the analog chain of the digital receiver.

[0005] It is recalled that in a digital receiver, the analog chain refers to the series of analog components and processes that manage the reception and conversion of the analog radio frequency (RF) signal into a digital format. This is the initial stage during which the incoming RF signal is conditioned and prepared for further digital processing.

[0006] An analog chain often includes the following components: - an antenna which is responsible for capturing the RF signal from the waves and converting it into an electrical signal; - a low noise amplifier (LNA) which amplifies the weak electrical signal received from the antenna while introducing minimal noise. - a band-pass filter that eliminates unwanted signals and noise outside the desired frequency range; - a mixer that combines the filtered RF signal with a local oscillator signal to perform a down-conversion of frequency: it produces an intermediate frequency (IF) signal, which is easier to process and digitize; - an intermediate frequency (IF) filter that eliminates unwanted signals and noise outside the desired frequency range, thus improving signal quality and isolating the specific frequency band of interest; - an automatic gain control (AGC) circuit that adjusts the gain of the received signal to maintain a constant signal strength; - an analog-to-digital converter (ADC) that converts the analog IF signal into a digital representation that can be processed by digital signal processing (DSP) components in the digital chain.

[0007] Once the analog chain has processed the signal and converted it into digital format, the digital chain performs the necessary processing on the signal (DSP). The analog chain thus plays an essential role in reception by guaranteeing the fidelity and quality of the received signal. It allows to minimize noise, filter unwanted frequencies and optimize the signal strength before digital processing takes place, which ultimately contributes to the overall performance of the digital receiver. The digital chain processes the signal for example by performing PSK-type phase demodulation or quadrature amplitude demodulation of 64QAM or higher, or any other type of phase and amplitude demodulation of a signal.

[0008] To meet these requirements at the analog chain level, it is necessary to resort to complex and expensive implementations. These implementations can be expensive in terms of their implementation: for example, to guarantee a minimum of distortions at the radio reception chain level, a known method consists of sorting and selecting electronic components. This sorting is often carried out manually, by testing one by one the electronic components to be used to manufacture the radio receiver. The additional cost incurred is therefore significant.

[0009] These implementations can be technologically complex, for example in terms of the configuration of the gain in the analog chain; in terms of filtering and equalization (use of static equalizers, which are mainly analog all-pass filters to compensate for the phase but not the amplitude and only in baseband); or in terms of physical implementation and design of the analog chain, requiring minimizing the length of the signal path and limiting sources of interference.

[0010] Document US10998926 proposes a solution for compensating for the flattening of the gain of a reception chain using a filter, the compensation being limited to the amplitude. This solution, however, requires the use of several signals and is complex to implement, since it requires working on a large number of points, which increases the computational cost on the one hand and the processing latency time in operational conditions, which is prohibitive for processing that must be carried out in real time.

[0011] It is therefore necessary to propose a solution for compensating for distortions in an analog reception chain which is simple and economical in terms of implementation, while responding to the problem of increasing the flow rates to be processed, and which is adapted to real-time constraints. 1. Summary of the invention#

[0012] To this end, the invention relates to a method for calculating static compensation coefficients of an analog reception chain, a method implemented by an electronic device (2) during a calibration phase of a reception device (3) comprising the analog reception chain and a digital compensation module implementing an FIR filter whose coefficients are the calculated static compensation coefficients, the method comprising:

[0013] - an estimation step (A01), using a predetermined calibration signal, of a frequency response H of the analog reception chain;

[0014] - a selection step (A02), using the frequency response H, of a band BP bandwidth on which the static compensation coefficients are calculated;

[0015] - a step of calculating (A03) the static compensation coefficients of the chain analog reception, in the time domain, as a function of the bandwidth BP and the inverse of the frequency response HBP over this bandwidth.

[0016] According to particular modes of implementation, the method comprises one or more of the following characteristics: - ; the estimation step (A01) of the frequency response H of the chain of analog reception includes: • a step of determining a reference wobble signal, in a frequency band of a predetermined width; • a step of obtaining a signal spectrum at the output of the analog reception chain, as a function of the reference wobble signal; • a step of calculating the frequency response H of the analog reception chain as a function of the signal spectrum at the output of the analog reception chain and the reference wobbled signal spectrum; - the step of selecting (A02) the bandwidth BP on which the static compensation coefficients are calculated comprises a step of selecting, within the frequency response H, a portion HBP corresponding to the portion of the frequency response H whose amplitude exceeds a predetermined amplitude value; - said amplitude value is predetermined in: • calculating the average value of the amplitude response of the frequency response H of the analog reception chain; • subtracting, from this average value of the amplitude response of the frequency response H of the analog reception chain, a predefined tolerance value; - the step of calculating (A03) the static compensation coefficients of the analog reception chain as a function of the bandwidth BP and the inverse of the frequency response HBP on this bandwidth comprises at least one iteration of a calculation of coefficients of a finite impulse response (FIR) filter running in the time domain as a function of the inverse Hi of the frequency response HBP on the bandwidth BP, said static compensation coefficients being constituted by the coefficients of the FIR filter; - the calculation of the coefficients of a finite impulse response (FIR) filter includes: • a step of obtaining a current number of NFFTk points from an inverse Fourier transform, N FFTk being a power of 2 between a minimum value and a maximum value; • with the number of points N FFTk current of the inverse Fourier transform ifft, a step of solving the equation:

[0017] hik{t) NFFTk)

[0018] with:

[0019] hïk^t} being the impulse response;

[0020] being the inverse response of the transfer function in the BP bandwidth;

[0021] this calculation delivering NFFTk values ​​of FIR filter coefficients; • a step of digital filtering of a predetermined signal using the FIR filter whose NFFn coefficients were obtained in the previous step; • a step of evaluating the quality of the FIR filter based on an EVM performance criterion; - the number of points N FFTk^i between 16 and 512.

[0022] The invention further relates to an electronic device (2) for calculating static compensation coefficients of an analog reception chain, a device intended to be used during a calibration phase of a reception device (3) comprising the analog reception chain and a compensation module. digital implementing an FIR filter whose coefficients are the calculated static compensation coefficients, the electronic device comprising: - means for estimating, using a predetermined calibration signal, a frequency response H of the analog reception chain; - means for selecting, using the frequency response H, a bandwidth BP on which the static compensation coefficients are calculated; - means for calculating the static compensation coefficients of the analog reception chain, in the time domain, as a function of the BP bandwidth and the inverse of the HBP frequency response over this bandwidth.

[0023] Furthermore, the invention relates to a computer program comprising software instructions which, when executed by a programmable electronic device, implement a method for calculating static compensation coefficients of an analog reception chain as defined above.

[0024] The invention finally relates to a reception device (3) comprising an analog reception chain and a digital distortion compensation module for the analog chain implementing an FIR filter, the digital compensation module being configured from coefficients calculated according to the method as defined above. 1. Brief Description of the Figures#

[0025] The invention will appear more clearly on reading the description which follows, given solely by way of non-limiting example, and made with reference to the drawings in which: - [Fig. 1] illustrates a system implemented to calibrate a receiver using a device for calculating the coefficients of a compensation filter; - [Fig.2] illustrates a device for calculating coefficients of a filter of compensation; - [Fig.3] explains the different steps implemented to calculate the correction filter coefficients; - [Fig.4] illustrates a wobbled signal of constant amplitude (between -1 and 1) and variable frequency over time; - [Fig.5] illustrates a comparison of the spectra of the input wobbled signal with the exit one; - [Fig.6] illustrates an amplitude response of a wobbled output signal from the analog chain, response obtained from a frequency response H of the analog chain; - [Fig.7] illustrates a phase response of a wobbled output signal from the analog chain, response obtained from a frequency response H of the analog chain; - [Fig.8] illustrates the number of points necessary (16) for the correction of a first analog channel in a first example situation; - [Fig.9] illustrates the number of points required (256) to correct a second analog channel in a second example situation; - [Fig. 10] illustrates the distribution of symbols for an output signal of an analog chain before correction by the method of the invention; and - [Fig. 11] illustrates the distribution of symbols for an output signal of an analog chain after correction by the method of the invention. 1. Description of an embodiment 1. Reminders of the principles# #

[0026] As indicated previously, an object of the invention is to carry out digital compensation for the distortions of an analog reception chain. This compensation, according to the invention, is simple to implement insofar as it does not result in any significant modification of the digital chain and it is carried out over a restricted bandwidth. It therefore requires little calculation and is compatible with real-time processing constraints.

[0027] Compensating for these distortions makes it possible to reduce the implementation cost by relaxing the linearity constraints of the filters in the analog chain. The solution implemented generally consists of inverting the amplitude response and / or the phase response (these responses being estimated during calibration) in the digital chain in order to correct the distortion of the overall analog chain and thus respect an imposed template (for example in terms of EVM - "error vector magnitude") and avoids resorting to more expensive methods such as component sorting.

[0028] According to the invention, the digital compensation is carried out using a compensation filter which is defined from static compensation coefficients. More particularly, the compensation filter is a finite impulse response (FIR) filter. The static compensation coefficients of the FIR filter are calculated only once for a current radio chain. The calculation of these coefficients is carried out in an initial calibration phase of the analog chain. This calculation can be carried out again during a subsequent recalibration of the chain (for example during maintenance). Thus, the static compensation coefficients are fixed once and for all at the end of the calibration. The advantage is that it is not necessary to modify the digital chain significantly, which is the case when it is desired to carry out dynamic compensation, in which the coefficients are automatically recalculated based on the received signal. The other advantage, as explained previously, is to limit not only the implementation cost but also the processing latency caused by this compensation in the receiver which implements the analog chain and the FIR filter.

[0029] According to the invention, these static compensation coefficients are calculated based on a frequency response of the entire reception chain. The objective is to compensate for the overall distortions of the analog reception chain, and not only those introduced by the analog filters, although the latter are mainly responsible for all the distortions.

[0030] More precisely, according to the invention, the distortions are compensated only in the bandwidth defined at -2dB from the central frequency. Then, the inverse response to be applied in the time domain is calculated. Indeed, a correction directly in the frequency domain would be simpler but more expensive to implement and therefore prohibitive. The inventors have, instead, carried out an estimation of an equivalent time filter, which is more easily implementable and therefore more economical for the digital chain.

[0031] [Fig.l] illustrates a system that can be implemented to calibrate a receiver 3 using the calculation device 2 for calculating coefficients of a compensation filter. For example, the calculation device 2 is implemented in a calibration phase and it receives the analog RX chain of the receiver 3. Signals are transmitted to this analog RX chain and transformed signals are analyzed by a network analyzer AR, in particular the output signals of the analog RX chain of the receiver 3 during calibration. The signals entering and leaving the analog RX chain are supplied to a calculation component CC for calculating the static coefficients of the compensation filter, which performs the calculation of these coefficients according to the method of the invention. When the coefficients of this filter are obtained, they are implemented in the filter of the receiver 3.

[0032] The receiver 3 comprises the analog RX chain, the digital filter which performs the compensation, using the components which are installed at the end of the calibration, so that the corrected signal can be processed by the rest of the digital chain in baseband.

[0033] [Fig.2] describes more precisely an electronic device for calculating static compensation coefficients of an analog reception chain. This device can be implemented in a calibration phase of a reception device comprising the analog reception chain and a digital compensation module.

[0034] In the example of [Fig.2], the electronic device for calculating static compensation coefficients 2 comprises an electronic memory unit 10, at at least one calculation processor 70 and an interface 80 for communication with remote devices, by a chosen communication protocol, for example a wired protocol and / or a radio communication protocol. The elements of the device 2 are adapted to communicate via a communication bus 90.

[0035] In the example of [Fig.2], a frequency response estimation module 20, a bandwidth determination module 40 and a static compensation coefficient calculation module 60 are each produced in the form of software, or a software brick, executable by the processor 70. The memory 10 of the electronic device for calculating static compensation coefficients 2 is then capable of storing frequency response estimation software 20, bandwidth determination software 40 and static compensation coefficient calculation software 60. The processor 70 is then capable of executing each of these software programs.

[0036] In a variant not shown, the frequency response estimation module 20, the bandwidth determination module 40 and the static compensation coefficient calculation module 60 are each produced in the form of a programmable logic component, such as an FPGA (Field Programmable Gate Array), or an integrated circuit, such as an ASIC (Application Specific Integrated Circuit).

[0037] When the electronic device for calculating static compensation coefficients 2 is produced in the form of one or more software programs, that is to say in the form of a computer program, also called a computer program product, it is also capable of being recorded on a medium, not shown, that can be read by a computer. The computer-readable medium is, for example, a medium capable of storing electronic instructions and of being coupled to a bus of a computer system. By way of example, the readable medium is an optical disk, a magneto-optical disk, a ROM memory, a RAM memory, any type of non-volatile memory (for example FLASH or NVRAM) or a magnetic card. A computer program comprising software instructions is then stored on the readable medium.

[0038] The method implemented for the calculation of these static compensation coefficients is described in relation to [Fig.3] and it comprises the following steps: - estimation A01, using a predetermined calibration signal SCP, of a frequency response H of the analog reception chain, delivering an amplitude response and / or a phase response; - selection A02, using the frequency response H, of a bandwidth BP on which the static compensation coefficients CSC will be calculated; - (iterative) calculation A03 of the static compensation coefficients of the analog reception chain, in the time domain, as a function of the BP bandwidth and the inverse of the HBP frequency response in this bandwidth: this is an inverse filter calculation in the bandwidth, in time allowing for example a minimization of the EVM, in an example of realization.

[0039] Once these compensation coefficients are static, they are saved in a memory and made accessible to the FIR filter implemented in the digital chain of the receiver being calibrated. This FIR filter will, subsequently, under operational conditions, filter the signal it receives using the calculated coefficients.

[0040] The FIR filter, calculated iteratively in the step of calculating the static compensation coefficients, represents a non-symmetrical response, i.e., non-real. Thus, it is not possible to apply it directly in baseband. The inventors therefore found it judicious to apply it either in intermediate frequency before descending to baseband or artificially on a low frequency sufficient to cover the filter band. We denote by f the artificial intermediate frequency such that / , >Bw / 2 and Bw represents the filter bandwidth.

[0041] In some implementation cases, however, it is not possible to operate on the artificial intermediate frequency f;. In this case, an alternative is to use a real filter (i.e. symmetrical) less suitable and less precise but nevertheless allowing gains in terms of performance.

[0042] To obtain the time response, the assumed known frequency response is used. This overall frequency response is reduced to its bandwidth and then inverted. Its impulse response is then deduced via inverse Fourier transform processing. This processing requires a certain number of points, ranging for example from 16 to 512, in order to be sufficiently precise.

[0043] It will be noted that one of the particularities of the method described above lies in its radio application which aims to optimize the particular performance criterion that is the EVM. Other equalization techniques are more studied but very oriented towards waveforms and propagation channel, and therefore are not adapted to the problem solved by the invention. 1. Implementation Example#

[0044] An example of implementation of the method previously described is presented below. 1. Estimation of the frequency response of the chain#

[0045] Firstly, as indicated previously, the device 2 carries out an estimation of the frequency response of the analog chain. In order to obtain this overall frequency response, it is necessary to calibrate the radio equipment 3 for which the coefficients of the FIR filter are calculated. This calibration is implemented by the electronic device 2 by injecting a so-called chirp waveform, i.e. a signal whose frequency varies over time. In doing so, the different frequencies of interest in the band used by the analog chain of the receiver 3 are excited. The electronic device 2 then recovers the signal at the output of the analog chain of the receiver 3 affected by the effect of the (only) distortions.

[0046] According to the invention, the wobbled signal c(t) which is implemented is defined as follows:

[0047] _ Aexp (j2pi(f1 + ) xt)

[0048] With fi the starting frequency of the wobbled signal and A / the frequency band covered. A signal of this type is exemplified in [Fig.4].

[0049] In this case, it is interesting to know the bandwidth to be estimated in order to determine the correct width of the wobbled signal. For a chain with global B-band filtering, we thus define A / = B and f _ with fc theo J ] J ^theo 2 the theoretical central frequency of the chain to be characterized. For example, the bandwidth to be estimated can be between 1.2 and 2 MHz, in a real case explained in relation to [Fig.4] and 5.

[0050] A comparison of the spectra of the input wobbled signal, denoted wobbled In, with that of the output, denoted wobbled Out, is exemplified in [Fig.5]. In this example of [Fig.5], we note the attenuations at the edge of the band which represent the filtering effects of the analog reception chain.

[0051] The following operation consists of dividing these two spectra to obtain the frequency response H of the chain. This frequency response includes both the phase and the amplitude:

[0052] ( \

[0053] In which: - fft is the fast fourier transform function defined as:

[0054] = ^F^x{nyj^Nm. avcc k, n = 0... NFFT-1 and NFFT the number of FFT points used which is a power of 2. In the calibration phase, there is no limit value on this number of points which must be chosen to obtain the most faithful representation possible of the frequency response; - S- [«] — S- (nT^ represents the discrete signal obtained by sampling at the rate f =}lTe of the time signal emitted at the input of the analog reception chain; $ 11n 1 = rcPæscntc 'c discrete signal obtained by sampling at the rate y — \ j Tg of the time signal received at the output of the

[0055]

[0056]

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[0058]

[0059]

[0060]

[0061]

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[0065] analog reception channel. Equipped with this frequency response H, it is possible to obtain the amplitude response and the phase response. [Fig.6] and 7 illustrate these two types of response (amplitude and phase) as an example. 1. Calculation of filter coefficients# Secondly, the electronic device 2 calculates the static compensation coefficients used in the digital domain in order to compensate for the distortions of the analog radio chain. These static compensation coefficients are determined from the frequency response H. To do this, we consider the transfer function of a following finite impulse response (FIR) filter, with a frequency response H: In which: - represents the delay of k samples; - bo, bi, b 2, ■■■> b M are the coefficients of the FIR filter; M is the order of the FIR filter. The order of the FIR filter indicates the number of coefficients N required for its implementation with N = M+l. The static compensation coefficients are, according to the invention, the coefficients of an FIR filter, and it is these FIR filter coefficients that we seek to determine, within the framework of this implementation. It is therefore necessary to calculate the different coefficients b 0, bj, b 2, ■■■> b M of such a filter. This calculation, however, is not carried out immediately from the frequency response H previously obtained, but from a portion HBP of it which should be determined, during the selection step A02, this portion being limited to a so-called correction bandwidth, within which one seeks to correct the distortions caused by the analog chain. 1. Determination of the correction bandwidth We select the bandwidth Bw of the filter defined at -2 dB of the amplitude value corresponding to a real central frequency noted fc. Indeed, this real central frequency fc can be (and often is) different from the theoretical central frequency of the filter, noted fctheo, precisely because of the dispersions of the analog filters that we wish to correct. To do this, we first consider the average value of the amplitude response in dB:

[0066]

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[0068]

[0069]

[0070]

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[0075]

[0076]

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[0082] As well as the overall amplitude response in dB: G,4 / ) = 201og, jHtd With ll the absolute value operator. We then select the response and associated frequencies that satisfy a certain tolerance with respect to the value average : HBP(f) = H(fgp) With fBP= f(GrfB(f)> H-tolerance) The tolerance value is for example set to 2dB. The center frequency is then calculated as the average frequency of this frequency range defined at 2 dB from the average amplitude value: fc= g.^^ In which: represents the last frequency of the frequency range f bp; - ffijo] represents the first frequency of the frequency range f Bp; Thus, at the end of this calculation step, we have the correction bandwidth and the central frequency of this correction bandwidth. 1. Calculation of coefficients minimizing the EVM# We denote by HBP the transfer function in the correction bandwidth. We then calculate the inverse response of the transfer function in the bandwidth. From this we deduce the impulse response that can be applied in the time domain. As stated previously, the objective is to calculate the coefficients of an FIR filter (and not an IIR) for simplicity and ease, particularly of implementation. The calculation of the coefficients of this FIR filter is implemented as follows: N fft ) With : - h^t) being the discrete impulse response in the time domain; - Hi(f) being the inverse response of the transfer function in the BP bandwidth; - ifft the inverse Fourier transform function defined as: iff{n, N,wNpprl aNmk number of FFT points used which is a power of 2.N FFT the number of points to use to calculate the impulse response.

[0083] The number of points N FFT of the inverse Fourier transform directly influences the resolution the estimation of the response, as well as their value. The number of FIR filter coefficients used is equal to the number of N FFT points of the inverse Fourier transform. The values ​​of the FIR filter coefficients used are equal to the resolution values ​​of the inverse Fourier transform.

[0084] Since the compensation is static, it is possible to establish the number of points required in the calibration phase in order to meet a certain EVM criterion. In the case of SAW type analog filters, this number of points can, for example, vary from 16 to 512 depending on the EVM performance criterion.

[0085] In other words, depending on the performance criterion, it can be determined that it will be necessary to calculate between, for example, 16 and 512 points of the inverse Fourier transform, or for example between 16 and 512 static compensation coefficients. According to the invention, the number of points of the inverse Fourier transform (i.e. the number of static compensation coefficients) is a power of 2. Furthermore, in order not to introduce a time delay which would be due to the use of a filter comprising too many coefficients (i.e. 2048 and higher for example), the number of points of the inverse Fourier transform is necessarily limited. Thus, the determination of the static compensation coefficients to be implemented in the filter for a current analog radio reception chain comprises at least one iteration of the following coefficient calculation steps, consisting of: Obtain a number of points N FFTk from the current inverse Fourier transform (i.e. 16, 32, etc.), N FFTk being a power of 2 between a minimum value and a maximum value (for example between 16 and 512), such that N FFTk = 2 k (k then being between 5 and 9); With the number of points N FFTk of the current inverse Fourier transform, solve:

[0086] = Npm)

[0087] this calculation delivers NFFTk FIR filter coefficient values; Digitally filter a predetermined signal, such as a modulated signal representative of the final application such as PSK or QAM modulation for example, using the FIR filter whose N FFn coefficients have been obtained in the previous step; Evaluate the quality of the filter based on an EVM performance criterion.

[0088] The iterations of the coefficient calculation steps are repeated until the EVM performance criterion is reached (for example to reach an EVM lower than 5%). When the performance criterion is reached, the calculation stops: we then have on the one hand the number of coefficients NFFna as well as the value of these coefficients by the resolution of the inverse Fourier transform.

[0089] In this exemplary embodiment, the performance criterion is defined as an EVM criterion. Other performance criteria could of course be used depending on requirements. This EVM performance criterion is defined, for a given constellation, with a given frame: 100901 ^=(4-4)^(^-¾) 2 100911 EVMt=lOoJ^ K y iayg - With P avg the average power of the constellation considered; - Ik the phase measurement of the kth symbol of the frame; - ïk the expected value in phase of the kth symbol of the frame; - Qk the quadrature measure of the kth symbol of the frame; - the expected quadrature value of the kth symbol of the frame.

[0092] Thus, the number of points required (i.e. the number and value of the coefficients) is determined during calibration as a function of the analog filter(s) that need to be corrected (and therefore as a function of the radio channel to which this or these filters relate). For example, in relation to Figures 8 and 9, for a first radio channel, to correctly correct the distortions, the number of points may be equal to 16 while for a second radio channel, the number of points may be equal to 256. More particularly, [Fig.8] illustrates that for a first radio channel, only 16 coefficients are required to fall below a threshold of 5% EVM. [Fig.9] illustrates that for a second radio channel, distinct from the first, no less than 256 coefficients are required to approach the threshold of 5% EVM.The number of coefficients calculated is therefore variable, and is a function of the radio chain and its uniqueness, in particular depending on the electronic components that compose it. Figures 10 and 11 illustrate the result obtained, in terms of EVM, using an FIR filter comprising coefficients calculated using the calculation method explained above. The advantage of this implementation is that we are able to accept a certain variability of the radio chain, variability which is compensated by the static filter, of finite length. 1. Application of the calculated filter coefficients

[0093] Equipped with the static filtering coefficients implemented in the FIR of the digital compensation module of the receiver, a correction is applied to the signal received. This correction is carried out under operational conditions, that is to say when using the radio receiver, more particularly, when using the reception device comprising the analog reception chain and the digital distortion compensation module of the analog chain, the digital module being configured from coefficients calculated during calibration.

[0094] However, in certain cases, in order to be able to apply this correction in baseband, it is necessary to transpose the impulse response of an intermediate frequency fi in order to be able to filter in the real domain:

[0095] hnco[n] = Zrjn] x exp (j2pi xf; xn) - is a discrete pulsed æPonsc serving as in-band correction base ; - hncoM is the impulse response ht[ n ] transposed to an intermediate frequency; - N represents the discrete samples in the time domain

[0096] This intermediate frequency fi is selected at the minimum necessary, i.e. at Qf, in order not to resort to excessive oversampling. In fact, to apply this correction, the sampling frequency (intermediate frequency fi) of the signal to which the correction is applied must be greater than or equal to the bandwidth of the filter:

[0097] f. > l- i.

[0098] We then apply this impulse response h nco by convolution to the signal to be corrected, itself shifted by this frequency fi:

[0099] SncJ»]= Soujn]x exp(j2pi xf;xn)

[0100] ç Ll_ ç Ll ^compbcol11] ^411] unco[u]

[0101] with: * the convolution operator.

[0102] We then go back down to baseband:

[0103] she n / -n • r \ SCOmp[n] = x exp( -j2pt x 1;xn)

[0104] The filtered signal is then processed by the next module in the digital chain.

Claims

Claims

1. Method for calculating static compensation coefficients of an analog reception chain, method implemented, by an electronic device (2), during a calibration phase of a reception device (3) comprising the analog reception chain and a digital compensation module implementing an FIR filter whose coefficients are the calculated static compensation coefficients, the method comprising: - a step of estimating (A01), using a predetermined calibration signal, a frequency response H of the analog reception chain; - a step of selecting (A02), using the frequency response H, a bandwidth BP on which the static compensation coefficients are calculated;- a calculation step (A03) of the static compensation coefficients of the analog reception chain, in the time domain, as a function of the bandwidth BP and the inverse of the frequency response HBP on this bandwidth.;

2. Method for calculating static compensation coefficients of an analog reception chain according to claim 1, characterized in that the step of estimating (A01) the frequency response H of the analog reception chain comprises: - a step of determining a reference wobbled signal, in a frequency band of a predetermined width; - a step of obtaining a signal spectrum at the output of the analog reception chain, as a function of the reference wobbled signal; - a step of calculating the frequency response H of the analog reception chain as a function of the signal spectrum at the output of the analog reception chain and the reference wobbled signal spectrum.

3. Method for calculating static compensation coefficients of an analog reception chain according to claim 1 or 2, characterized in that the step of selecting (A02) the bandwidth BP on which the static compensation coefficients are calculated comprises a step of selecting, within the frequency response H, a portion HBP corresponding to the portion of the frequency response H whose amplitude exceeds a predetermined amplitude value.

4. Method for calculating static compensation coefficients of an analog reception chain according to claim 3, characterized in that said amplitude value is predetermined by: - ​​calculating the average value of the amplitude response of the frequency response H of the analog reception chain; - subtracting, from this average value of the amplitude response of the frequency response H of the analog reception chain, a predefined tolerance value.

5. Method for calculating static compensation coefficients of an analog reception chain according to any one of the preceding claims, characterized in that the step of calculating (A03) the static compensation coefficients of the analog reception chain as a function of the bandwidth BP and the inverse of the frequency response HBp on this bandwidth comprises at least one iteration of a calculation of coefficients of a finite impulse response (FIR) filter running in the time domain as a function of the inverse H; of the frequency response Hbp on the bandwidth BP, said static compensation coefficients being constituted by the coefficients of the FIR filter.

6. Method for calculating static compensation coefficients of an analog reception chain according to claim 5, characterized in that the calculation of the coefficients of a finite impulse response (FIR) filter comprises: - a step of obtaining a current number of points NFFTk of an inverse Fourier transform, NFFTk being a power of 2 between a minimum value and a maximum value; - with the current number of points N FFTk of the inverse Fourier transform ifft, a step of solving the equation:

7.

8.

9. hikW ^IF Tk ) with : hik(t ) being the impulse response; being the inverse response of the transfer function in the BP bandwidth; this calculation delivering NFFTk values ​​of FIR filter coefficients; - a step of digital filtering of a predetermined signal using the FIR filter whose N FFn coefficients were obtained in the previous step; - a step of evaluating the quality of the FIR filter based on an EVM performance criterion. Method for calculating static compensation coefficients of an analog reception chain according to claim 6, characterized in that the number of points N fft^I is between 16 and 512. Electronic device (2) for calculating static compensation coefficients of an analog reception chain, device intended to be used during a calibration phase of a reception device (3) comprising the analog reception chain and a digital compensation module implementing an FIR filter whose coefficients are the calculated static compensation coefficients, the electronic device comprising: - means for estimating, using a predetermined calibration signal, a frequency response H of the analog reception chain; - means for selecting, using the frequency response H, a bandwidth BP on which the static compensation coefficients are calculated; - means for calculating the static compensation coefficients of the analog reception chain, in the time domain, as a function of the BP bandwidth and the inverse of the HBP frequency response over this bandwidth. Computer program comprising software instructions which, when executed by a programmable electronic device, implement a method for calculating static compensation coefficients of an analog reception chain in accordance with any one of claims 1 to 7.

10. Reception device (3) comprising an analog reception chain and a digital distortion compensation module for the analog chain implementing an FIR filter, the digital compensation module being configured from coefficients calculated according to the method of any one of claims 1 to 7.

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