Method and device for measuring quality of signal shape

FIELD: method and device for measuring quality of signal shape.

SUBSTANCE: real signal, representing shape of signal, divided on separate channels by time and codes, is produced, for example, by means of standard communication system for high speed data transfer. Controlling-measuring equipment produces ideal signal shape, matching real signal shape. This equipment produces estimate of shifts between parameters of real signal shape and ideal signal shape, then performs estimation of different measurements of quality of signal shape using quality measurements of compensated real shape of signal. Examples of processing real signal shape and appropriate ideal signal shape by means of controlling-measuring equipment are given. Provided method and devices can be utilized with any shape of signal, separated on channels by time and codes, not depending on equipment, which produces signal shape.

EFFECT: increased precision of measurement of signals shape quality, which are separated on channels in temporal area and code area.

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The technical field

The present invention relates to quality assurance. More specifically, the present invention relates to a method and apparatus for measuring the quality of the signal.

Prior art

In recent years we have developed a communication system that allows you to transmit signals from the station, the message source to differing essentially the destination station. When the signal transmission from the station, the source of the message via the communication signal is first shaped in the form of providing efficient transmission over a communications line. Used communication line contains medium on which the signal is transmitted. Conversion or modulation signal includes modifying a parameter of the carrier signal in accordance with the signal in such a way that the spectrum of the resulting modulated carrier was limited to within the bandwidth of the communication line. At the destination station, the original signal is reproduced from a version of the modulated carrier received over the communication line. Such reproduction is achieved, in General, using the reversible process of modulation used by the station of origin of the message.

Modulation also facilitates multiple access, i.e. simultaneous transmission and/or reception of multiple signals over a common communication line. Communication system with multiple to blunt often include many remote subscriber modules, requiring periodic maintenance with a relatively short duration, and not continuous access to a shared communication line. In the technique known several methods multistation or multiple access, such as multiple access with time division multiplexing (TDMA), multiple access frequency division multiple access (FDMA) and amplitude modulation (AM). Another type of multiple access method is a multiple access code division multiple access (CDMA) system with the extension of the spectrum which corresponds to the "Standard compatibility "mobile station - base station" TIA/BIA/IS-95 for dual-mode wideband cellular systems with expansion of the range", which is hereinafter referred to as the standard IS-95. Application of methods in CDMA communication system with multiple access are disclosed in U.S. patent No. 4901307 "communication System with multiple access and widening of the range using satellite or terrestrial repeaters," and in U.S. patent No. 5103459 "System and method for producing waveforms in a cellular telephone system, CDMA, each of which belongs to the authors of the present invention and is incorporated into this description by reference.

In Fig. 1 shows an ideal waveform 100 variants of implementation of the communication system is a code division in accordance with the standard IS-95. This document is the waveform is used for the display, representation or visualization of the signal pulse or transition. The idealized waveform 100 includes parallel channels 102, which differ from each other by code coverage. Code coverage in the communication system according to the standard IS-95 contains Walsh codes. The ideal waveform 100 is then subjected to quadrature expanding range of filtration in the primary frequency band and the conversion frequency relative to the carrier frequency. The resulting modulated waveform 100, is expressed as:

where ωwithis the nominal frequency of the carrier waveform, i is the index of summation code channel, Ri(t) is the complex envelope of the ideal i-th code channel.

Equipment, such as the transmitter of the communication system is a code division that produces the actual shape of the signal x(t), which differs from the ideal waveform. This is valid waveform x(t) is expressed as:

where bi- ideal amplitude waveform with respect to the ideal waveform for the ith code channel, τithe time offset of the ideal form of the signal in relation to the ideal waveform for the ith code channel, Δω - offset circular frequency signal, θi- the phase shift of the ideal form rings the La relative to the ideal waveform for the ith code channel, Ei(t) is the complex envelope of the error (deviation from ideal) of a valid signal for the ith code channel.

The difference between the ideal waveform s(t) and the actual shape of the signal x(t) is measured as maximum deviation frequency tolerance time the pilot signal and compatibility in the form of a signal. One way to perform this measurement is to determine the accuracy of the modulation, which is defined as the proportion of valid power waveform x(t), which correlates with the ideal waveform s(t), when the transmitter modulation is performed using a code channel. The accuracy of the modulation can be expressed as:

where T1- the beginning of the integration period, T2the end of the integration period.

For systems with discrete time, where s(t) and x(t) are discrete signals at the points tkideal discretization, equation (3) can be written in the form

where Xk= x[k] = x(tk) - k-I value of the actual waveform, Sk = s[k] = s(tk), the corresponding k-I value of the ideal waveform.

Communication system with multiple access allows you to transmit voice signals and/or data. An example of a communication system, a transmitting voice signals and data is C the subject, the relevant standard IS-95, which accurately determines the transmission voice signal and data communication lines. The data transmission method in the code channel frames of predetermined size are described in detail in U.S. patent No. 5504773 "Method and apparatus for the formatting of data for transmission", which belongs to the authors of the present invention and is incorporated into this description by reference. In accordance with the standard IS-95, the data or voice signals are divided into code channel frames with a duration of 20 milliseconds with data rates up to 14.4 kbit/s Additional examples of communication systems that transmit voice signals and the data contain a communication system corresponding to the "a Joint project of the 3rd generation (3GPP)and embodied in a set of documents that includes documents No. 3G TS 25.211, 3G TS 25.212, 3G TS 25.213, and 3G TS 25.214 (the standard W-MARK) or Standard physical layer TR-45.5 for systems with the expansion of the range MDRC-2000" (standard IS-2000). Such a communication system using a waveform similar to that which was described above.

Recently developed system of communication intended only for the transmission of data at high speed (WSPD (HDR)). Such a communication system is disclosed in pending application No. 08/963386 "Method and apparatus for transmitting packet data at high speed from 11.03.1997, which belong to IIT the authors of the present invention and included in the description by reference. Communication system SPD defines the set of data rates ranging from 38.4 kbps to 2.4 Mbps, with which the calling terminal (access point, AP) may send data packets to the receiving terminal (terminal access, TD (at)). In the system WSPD used form of signal channels, which differ in the time domain and code domain.

In Fig. 2 shows such a waveform 200, modeled on the basis of the signal forms a straight line above WSPD. The waveform 200 is determined based on the frame 202. (Of Fig. 2 shows only the frames 202a, 202b, 202c.) According to one embodiments, the frame contains 16 time slots 204, each time slot 204 has a length of 2048 chips ("elementary signals")corresponding to the length of slot 1,67 milliseconds and, therefore, the duration of the frame 26,67 MS. Each slot 204 is divided into two polulate 204a, 204b, and the pilot-packages 206a, 206b are transmitted within each polulate 204a, 204b. According to one of embodiments, each pilot service 206a, 206b has a length of 96 chips and concentrated in the middle of their associated polulate 204a, 204b. Pilot-packages 206a, 206b contain the signal of the pilot channel, protected by a coating Walsh with index 0. The pilot channel is used for synchronization purposes. Direct channel 208 controls the medium access (MAC (MAC)) generates two package is 208a and two packages 208b, each of which has a length of 64 chips. Packages 208a, 208b DCP passed directly before sending the pilot-packages 206a, 206b each slot 204 and after transmission. According to one embodiments, the DCP consists of up to 63 code channels that are orthogonal protected 64 hex codes Walsh. Each code channel is identified by the index UDS, which has a value between 0 and 63 and identifies a unique 64-ranks coating Walsh. Indexes forward link control medium access DCP 0 and 1 reserve. Reverse link power control (CUM (RFC)is used to adjust the power signal return line connection for each subscriber station. Reverse link power control AFB assigned to one of the existing DCP index DCP 5-63. The DCP to DCP index 4 is used for the reverse channel activity (OA (RA)), which performs flow control on the reverse channel traffic. Channel traffic straight line and payload control channel is sent in the remaining portions 210a of the first polulate 204a and the remaining parts 210b of the second polulate 204b. Direct channel and traffic channel data management code, scrambling and alternating. As needed, the data is interleaved modulate, repeat and perforined. Then the resulting sequence of symbols of the modulation dealt flexiroll for to form 16 pairs (in-phase and shifted in phase by 90°) parallel threads. Each of the parallel threads close separate 16-ranks coating Walsh, producing visible on the codes of the channel 212.

The ideal waveform 200 is then quadrature extend along the spectrum, filtered in the primary frequency band and transform with increasing frequency on the carrier frequency. The resulting modulated waveform 200 is expressed in the form:

where ωwithis the nominal frequency of the carrier waveform, i(t) is the index of the coded channels. The index depends on the time, as the number of code channels varies over time, and Ri(t) is the complex envelope of the ideal i-th code channel, is represented as:

where aithe amplitude of the ith code channel, g(t) is the unit impulse response of the filter in the transmission in the primary frequency band, φi,kthe phase of the k-th chip of the ith code channel, in a discrete period of time tk= kTc, Tc- the duration of a chip.

The transmitter of the communication system WSPD produces the actual shape of the signal x(t)is represented as:

where bi- ideal amplitude waveform relative to an ideal waveform DL is the ith code channel, τithe time - shift the ideal form of the signal relative to the ideal waveform for the ith code channel, Δω - shift the circular frequency of the signal θi- ideal phase shift of the signal relative to the ideal waveform for the ith code channel, Ei(t) is the complex envelope of the error (deviation from the ideal waveform) of the actual signal transmission for the i-th code channel.

Based on a complex time domain and separation channels in the code domain waveform 200, the ways to measure the quality of the signal based on the formation of channels in the code domain, are unsuitable. Therefore, in the technique there is a need for method and device for measuring the quality of waveforms for waveforms that are divided into channels in the time domain and code domain.

Disclosure of inventions

The present invention is the task of developing a new method and device to measure the quality of the waveform. In accordance with the method to produce the real signal representing a waveform divided by the channels in the time domain and code domain. Such a valid waveform is possible to produce, for example, using the communication system. Instrumentation produces the ideal waveform, the corresponding actual signal form. Instrumentation then develops criteria for evaluating the displacement between the actual waveform and the ideal waveform and uses the offsets to compensate for the actual waveform. In one embodiment, the implementation is assessed the full precision of the modulation in accordance with compensated ideal waveform and the ideal waveform.

In another embodiment, the estimated modulation accuracy for a specific channel time division signal forms. Offset valid waveform is processed to implement specific channel by time division. According to one embodiments, the processing step includes assigning the compensated actual signal whose value is zero in intervals where elsewhere the specific channel time division is defined and nonzero. According to another variant implementation, the processing step includes multiplying the compensated actual signal form to a function with a value that differs from zero in intervals where the specific channel time division defined, and in other cases equal to zero. In one embodiment, the ideal waveform is treated the same way. In another embodiment, the imp is in immediately produce the ideal waveform, containing a specific channel by time division. The modulation accuracy for a specific channel time division assessed in accordance with the processed compensated actual signal form, and processed the ideal waveform.

In another embodiment, the estimation of the coefficients of the power code domain for a particular code channel. Specific channel by time division, which contains a specific code channel with compensated actual signal waveform obtained in accordance with the above-described method. In one implementation of the ideal waveform is treated the same way. In another implementation immediately produce the ideal waveform containing a specific code channel specific channel by time division. The modulation accuracy for a specific channel time division shall be assessed in accordance with the processed compensated actual signal form, and processed the ideal waveform.

Brief description of drawings

The features, objectives and advantages of the present invention is further illustrated by the description of its variants, with reference to the figures of the accompanying drawings, in which:

Fig. 1 depicts an idealized waveform of the communication system is a code section is of;

Fig. 2 depicts an idealized waveform communication systems high speed data (SPD) and

Fig. 3 depicts the concept of a device with the ability to implement quality measurement waveform in accordance with the principles of the present invention.

A detailed description of the preferred embodiments

In Fig. 3 shows the concept of the device with the ability to implement quality measurement waveforms for waveforms that are divided into channels in the time domain and code domain, such as the exemplary waveform 200 (Fig. 2).

In one embodiment, the real signal x(t) (representing the waveform 200 shown in Fig. 2) is entered in block 302 compensation. Block 302 compensation also allows to estimate the offset of the actual waveform x(t) with respect to the ideal form of the signal s(t)coming from block 304 optimization. Block 302 compensation uses estimates of the bias to obtain the compensated waveform y(t). Compensated waveform y(t) is supplied to the block 306 conversion with decreasing frequency. The signal converted to a lower frequency, then served in the additional block 308 sampling. Discretiona the shape of the signal z[k] is supplied in the optional block 310 convert the baseband frequency. The output signal z[k] of the optional block 310 converting baseband frequencies is given to block 312 processing.

In accordance with one variant of implementation, the ideal waveform s(t) is generated using a generator 314 signals. The ideal signal s(t) is supplied to the additional block 316 sampling. Discrete signal s[k] is served in the additional block 318 convert the baseband frequency. The output signal r[k] of the additional block 318 convert the baseband frequency is served in block 312 processing. In another embodiment, the generator 314 generates signals directly in digital signal r[k]. Therefore, in this embodiment, there is no need to block 316 sampling and additional block 318 convert the baseband frequency.

Block 312, the processing uses the signal z[k] and r[k] to calculate the characteristics of the waveforms.

As was shown above, the actual shape of the signal x(t) will be offset from the ideal waveform s(t) in terms of frequency, time and phase. A measure of the quality of the waveform is determined for the best alignment within the real waveform x(t), which is offset from the ideal waveform s(t). Therefore, a measure of the quality of the signal is evaluated for a variety of combinations of shifts in frequency, time and phase, the maximum of these estimates is taken as the criterion of quality. Function block 304 optimization is the button to produce many combinations of shifts in frequency, time and phase.

The function of block 302 of compensation is to operate the waveform x(t) to obtain the compensated waveform y(t)represented by equation (7):

where- estimation of the shift of the circular frequency of the signal x(t) with respect to the signal s(t)- evaluation of the time shift of the signal x(t) with respect to the signal s(t)- evaluation of the phase shift of the signal x(t) with respect to the signal s(t).

Parameters,andserved in block 302 compensated with block 304 optimization.

As was shown above, the shape of the signal x(t) was transformed with increasing frequency at the carrier frequency designation unit 306 conversion with decreasing frequency is to be converted with decreasing frequency offset waveform y(t) in the form of a signal z(t) baseband frequency.

In one embodiment, the implementation of additional block 308 sampling creates a discrete version of z[k] of the signal z(t) by sampling the waveform z(t) at the points tkthe ideal sample rate:

In another embodiment, an additional block 308 sampling is absent, the discretization is performed using block 312 processing after converting the base band frequency.

As shown previously, before sending the waveform 200 is filtered in the primary frequency band. Therefore, additional block 310 converting base band frequency is used to remove inter-symbol interference (MI (ISI), which makes the transmitter filter. In order to perform this operation, the transfer function block 310 conversion of the primary frequency band is back and a complex conjugate of the transfer function of an ideal transmitter filter.

Block 312 processing processes the signal z[k] and r[k] to perform the desired measurement quality waveform, as will be described in detail below. According to one of embodiments, when there is no additional block 308 sampling unit 312 processing creates a discrete version of z[k] of the signal z(t) by sampling the signal z(t) at the points tkthe ideal sampling rate in accordance with equation (9).

When considering the above device specialists can change the block diagrams for different representations of the waveforms x(t) and s(t). For example, if the waveform x(t) is represented in the form of a signal baseband frequencies in the digital domain, block 306 conversion with decreasing frequency and additional block 308 sampling may be missing. In addition, if the shape of the signal x(t) does not need to be filtered, can not the AMB additional block 310 convert the baseband frequency. In addition, experts can modify the block diagram in accordance with the type of measurement. For example, if you have installed the influence of the main filter bandwidth, the blocks 310 and 318 conversion baseband frequencies could not be used, therefore, in block 312, the processing was applied would be the ideal signal shape and signal the ideal form of blocks 308 and 316 sample rate.

The measurement of the modulation accuracy

The modulation accuracy is defined as the proportion of power in the real signal z[k], which correlates with the ideal signal r[k], when the transmitter is modulated at least one channel in the signal.

Full modulation accuracy is defined as the proportion of power in the real signal z[k], which correlates with the ideal signal r[k], when the transmitter is modulated by all channels in the signal. In one of the embodiments of the communication system of high-speed data transfer these channels include a pilot channel, OVC and direct trafc channel or control. First the overall accuracy of the modulation is determined as follows:

where ρtotal-1first the overall accuracy of the modulation, j is the index denoting the elementary element of waveform, N is the limit of the summation, indicating the number of elementary elements k in the CEN, denoting the discrete basic element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k= z [M(j-1)+k] - k-I readings in the j-th elementary element of the real waveform, and Rj,k= r[M(j-1)+k] - k-I readings in the j-th elementary element of the real waveform.

Basic element is defined as the minimum interval of the waveform that defines the overall structure of the channel. The value of the limit of summation N is chosen so that the noise variance of the measurements was below the required value.

Applying equation (10) to form signal 200 a direct line of communication system WSPD, the elementary item is proslot, therefore, the limit of summation M = 1024. The first value of z(t1) takes place in the first chip polulate, and the final value of z(t1024N) takes place in the last chip polulate. The value of the limit of summation N is equal to at least 2.

First the overall accuracy of the modulation does not explain the possible lack of continuity of the shape parameters of the signal at the boundaries of the elementary elements. Therefore, the second overall accuracy of the modulation is determined as follows:

where ρGeneral 2second overall accuracy modulation, j is the index denoting the elementary form element signal, N - prés who ate summation, denoting the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k= z[(M++1)·(j-1)+k] - k-I readings in the j-th elementary element of the real waveform, Rj,k= r[(M++1)·(j-1)+k] - k-I readings in the j-th elementary element of the real waveform.

Applying equation (11) to the waveform 200 direct lines of communication systems high-speed data transmission, the elementary item is proslot, therefore, the limit of summation M = 1024. The first value of z(t531) takes place in 513-m chip polulate, and the final value of z(t1536N) takes place in 513-m chip last polulate. The value of the limit of summation N is equal to at least 2.

The accuracy of the modulation of the channel by time division (Rusina) is defined as the proportion of power in the real signal z[k], which correlates with the ideal signal r[k], when the transmitter is modulated specific Ukanlos waveform. According to one of embodiments of the system WSPD high-speed data transmission channels include a pilot channel, direct channel access control environment OVC and direct trafc channel or control. The accuracy of the modulation is Canola is defined as follows:

where ρRusinol- accuracy modulation for channel separation by time, j is the index denoting the elementary element of waveform, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k= z[(M(j-1)+k] - k-I readings in the j-th elementary element of real Rusinol, Rj,k= r[(M(j-1)+k] - k-I readings in the j-th elementary ideal element Rusinol.

The concept of processing the real shape of the signal z[k] and the ideal form of the signal r[k] for specific Rusinol described below. The function gRusinolis defined as:

where mp≤ (k modL ≤ mp+1for p = 1,2,...n specifies the interval, where the waveform is not equal to zero for a particular Rusinol, L is the interval elementary element of the signal z[k].

Then the real shape of the signal z[k] and the ideal shape of the signal r[k] is multiplied by the function gRusinol[k] to generate specific Rusanov:

Specialists will be clear that the implementation of the concept may vary. According to one of embodiments, the processing is implemented in the form of multiplying the waveform function with the value, the cat is the second non-zero in the intervals, where specific channel by time division, and zero in all other cases. In another embodiment, the processing includes assigning the form of a signal value that is not zero in intervals where specific channel by time division, and zero in all other cases. In yet another embodiment, the processor implements the equation (12), configured to transfer the internal summation as follows:

where mp≤ (k mod L ≤ mp+1for p = 1,2,...n specifies the intervals at which the waveform is not equal to zero for a particular Rusinol, L is the interval elementary element signal z[k] and r[k].

Measurement code domain

Code domain power is defined as the share of power of the signal z(tk), which correlates with each code channel Ri(tk)when the transmitter is modulated in accordance with the known sequence of code symbols. The concept of processing waveform for each code channel Ri(tk) described below. In the beginning a specific Rusinol containing each code channel Ri(tk), using any of the above ways. For example, equation (13) is used to obtain fu the functions g Rusinol[k] for specific Rusinol. The function gRusinol[k] is then used to impact on the real shape of the signal z[k] and the i-th code channel Ri(tkideal signal r[k] in order to obtain waveforms:

Odds ρRusinol,ipower code quality waveform for a particular Rusinol then determine for each code channel Ri(tkas follows:

where ρMrvan,i- coefficient code for channel separation in time, identified by the index Mrvan, and code channel Ri(tk)identified by index i, w1the first code channel to channel MRI with time division, wvthe last code channel to channel diameter divided by time, j is the index denoting the elementary element of waveform, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal R'i,j,k=R'i[(M(j-1)+k] - k-I readings in the j-th elementary element of the i-th code channel of the ideal signal.

N the example, applying the above method for assessing ρUDS,iwaveform 200 direct line of communication system WSPD, elementary element is equal to polulate, therefore, the limit of summation M = 1024. From equation (13) and Fig. 2 follows:

where (k mod 1024) = 1 occurs at the first chip of each polulate. Then from equation (16) follows:

The following factors ρUDS,ipower code domain identify for a channel by using equation (17):

The value of N for measuring ρUDS,ii ≠ 4 was determined to be equal to at least 16. The first value of z(t1) takes place in the first chip polulate, the final value of z(t1024N) takes place in the last chip polulate.

Specialists will be clear that various listed as examples of logical blocks, modules, circuits, and steps of the algorithm described in conjunction with the implementation disclosed in the description may be implemented as electronic hardware, software, computers, or combinations thereof. Various examples of the component blocks, modules, circuits, and steps have been described generally in terms of their functionality. The implementation of functionality in the form of hardware or software is mnogo security depends on the particular application and design constraints, imposed on the whole system. The competence of the specialist is the ability to identify the interchangeability of hardware and software in this case and implement the described functionality for each specific application.

For example, the various illustrated logical blocks, modules, circuits, and steps of the algorithm described in conjunction with options for implementation disclosed herein, may be implemented or performed with a processor for digital signal processing (PCOS (DSP)), a specialized integrated circuit (ICI (ASIC)), user-programmable gate arrays (PPWM (FPGA)or other programmable logic devices, discrete gates or transistor logic, discrete hardware components, such as registers and device FIFO type, the processor executing a set of firmware commands, any known programmable software module and a processor, or any combination thereof. As processor can be used mainly microprocessor, but also any known processor, controller, microcontroller, or any state machine. A software module may reside in RAM memory, flash memory, RAM, ROM, registers, hard on the claim, removable disk, CD-ROM (CD-ROM) or any other form known in the art media. Specialists will be clear that the data, instructions, commands, signals, bits, symbols, and chips mentioned in the description, mainly represent the voltages, currents, electromagnetic waves, magnetic fields or particles, optical fields or particles, or any combination of them.

The above description of the preferred embodiments using communication systems that exemplify quality measurement waveform, allows any specialist to make or use the present invention. Various modifications of the embodiments will be clear to the experts, and the generic principles defined in the description, can be applied to other variants of implementation without the use of inventive activity. In particular, the experts will be clear that the disclosed generic principles apply equally to any form of signal, regardless of the equipment that generates the waveform. Thus, the present invention should not be limited to the above variant implementation, and represents the most broad concept set forth in the description.

1. The method of measuring the quality of a waveform, comprising stages, according to which

form the plural the creation shifts the parameters of the real signal relative to the ideal signal,

compensated real signal using multiple shifts for forming compensated real signal,

filter the compensated actual signal to obtain a filtered signal,

change ideal signal so that it matches filtered signal to obtain a modified signal, and

measure the quality of a waveform in accordance with the modified ideal signal and the filtered signal.

2. The method according to claim 1, whereby the step of forming multiple shifts form a shift in frequency, time shift and phase shift.

3. The method according to claim 1, whereby the phase compensation of the real signal with multiple shifts produce compensation in accordance with the following equation:

where y(t) - compensated real signal, x(t) is a real signal, t is time, j is the imaginary unit,the shift in frequency,- time shift,- phase shift.

4. The method according to claim 1, whereby the step of filtering assign compensated real signal value of zero in the filtered intervals and different from zero in the other intervals.

5. The method according to claim 4, whereby the step of filtering Priva who provide the compensated actual signal value, different from zero at all elementary element of the real signal.

6. The method according to claim 4, whereby the step of assigning the compensated actual signal values determine the function with a value of zero in the filtered intervals and different from zero in the other intervals, and multiply compensated real signal function.

7. The method according to claim 6, whereby the step of determining the function define a function with values other than zero for all elementary element of the real signal.

8. The method according to claim 1, whereby the step of modifying the ideal signal form the modified ideal signal that is zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals.

9. The method according to claim 1, whereby the step of modifying the ideal signal ideal signal is assigned a value of zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals.

10. The method according to claim 9, whereby the step of assigning an ideal signal of a certain value to define the function with a value that is zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals, and multiply compensated real signal h is a function.

11. The method according to claim 5, whereby the step of determining the quality of the waveform first calculate the overall accuracy of the modulation.

12. The method according to claim 11, whereby the first modulation accuracy is calculated in accordance with the following equation:

where ρtotal-1first the overall accuracy of the modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and Rj,k=r[M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

13. The method according to claim 11, whereby additionally calculate the second overall accuracy modulation.

14. The method according to item 13, according to which the second modulation accuracy is calculated in accordance with the following equation:

where ρGeneral 2- second accuracy modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, indicating the number of thesis the pet in the unit element, - k-I readings in the j-th elementary element of the filtered signal and- k-I readings in the j-th elementary ideal element signal.

15. The method according to claim 4, whereby the step of determining the quality of the waveform calculate accuracy modulation for channel separation by time.

16. The method according to item 15, whereby the accuracy of modulation for channel separation in time is calculated in accordance with the following equation:

where ρMrvan- accuracy modulation for channel Mrvan with time division, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and Rj,k=r[(M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

17. The method according to claim 4, whereby the step of defining quality measurement waveform calculate the coefficients of the power code domain.

18. The method according to 17, according to which the power factors of the code field is calculated in accordance with the following equation:

where ρMrvan,i- coefficient code domain channel Mrvan with time division and code channel i, w1the first code channel to channel Mrvan with time division, wvthe last code channel to channel Mrvan with time division, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and R'i,j,k=R'i[(M(j-1)+k] - k-I readings in the j-th elementary element of the i-th code channel of the ideal signal.

19. Device for measuring the quality of the waveform containing

the first means is arranged to provide a variety of shifts of the parameters of the real signal relative to the ideal signal,

the second means is arranged to compensate the real signal using multiple shifts with getting compensated real signal,

the third means is configured to filter the compensated actual signal to obtain a filtered signal,

the fourth tool, made the second with the possibility of changing the ideal signal, the corresponding filtered signal to obtain a modified signal, and

the fifth means is arranged to measure the quality of a waveform in accordance with the modified ideal signal and the filtered signal.

20. The device according to claim 19, in which the first means, second means, third means, the fourth means and the fifth means contain the instrumentation.

21. The device according to claim 19, in which the first means is formed with a possibility of providing a variety of shifts to ensure a shift in frequency, time shift and phase shift.

22. The device according to claim 19, in which the second means is formed with a possibility of real compensation signal by using a variety of shifts to estimate the following equation:

where y(t) - compensated real signal, x(t) is a real signal, t is time, j is the imaginary unit,the shift in frequency,- time shift,- phase shift.

23. The device according to claim 19, in which the third means is configured to filter for assigning the compensated actual signal values of zero in the filtered intervals and different from zero in the other intervals.

24. The device according to item 23, in which the third means is configured to filter for assigning the compensated actual signal values, different from zero at all elementary element of the real signal.

25. The device according to item 23, in which the third means is arranged to assign compensated real signal threshold value to define a function with a value of zero in the filtered intervals and different from zero in the other intervals, and for multiplying the compensated actual signal to the function.

26. The device according A.25, in which the third means is arranged to define a function with values other than zero for all elementary element of the real signal.

27. The device according to claim 19, in which the fourth means is configured to change the ideal signal to obtain the modified ideal signal to have a value of zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals.

28. The device according to claim 19, in which the fourth means is configured to change the ideal signal assignment of the ideal signal value that is zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals.

29. The device according to p, in which the fourth means is arranged to assign an ideal signal of a certain value in order to determine the possible functions with the specified value that is zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals, and multiplying the compensated actual signal to the specified function.

30. The device according to paragraph 24, in which the fifth means is arranged to determine the quality of the waveform to calculate the first overall accuracy modulation.

31. The device according to item 30, in which the fifth means is configured to calculate the first modulation accuracy for estimating the following equation:

where ρtotal-1first the overall accuracy of the modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[M(j-1)+k] -k-I readings in the j-th elementary element of the filtered signal, and Rj,k=r[M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

32. The device according to item 30, in which the fifth means is configured to calculate a second total accuracy modulation.

33. The device according to p, in which the fifth means is configured to calculate the second modulation accuracy for estimating the following equation:

where ρGeneral 2second overall accuracy modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element,- k-I readings in the j-th elementary element of the filtered signal, and- k-I readings in the j-th elementary ideal element signal.

34. The device according to item 23, in which the fifth means is arranged to determine the quality of the waveform to determine the accuracy of modulation for the channel by time division.

35. The device according to clause 34, in which the fifth means is arranged to determine the accuracy of modulation for the channel by time division to estimate the following equation:

where ρMrvan- accuracy modulation for channel separation by time, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in j-m elementar the m element of the filtered signal, and Rj,k=r[(M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

36. The device according to item 23, in which the fifth means is arranged to determine the parameters of quality measurement waveform and calculates the power factors of the code area.

37. The device according to p, in which the fifth means is configured to calculate coefficients of the power code domain to estimate the following equation:

where ρMrvan,i- coefficient code domain channel Mrvan with time division and code channel i, w1the first code channel to channel Mrvan with time division, wvthe last code channel to channel Mrvan with time division, j is the index denoting the elementary element signals, the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and R'i,j,k=R'i[(M(j-1)+k] - k-I readings in the j-th elementary element of the i-th code channel of the ideal signal.

38. Device for measuring the quality of the waveform containing

processor and

is the eh information associated with the processor and containing a set of commands executable by the processor to

provide a variety of shifts of the parameters of the real signal relative to the ideal signal,

the real compensation signal by using a variety of shifts to generate compensated real signal,

filtering the compensated actual signal to generate a filtered signal,

change ideal signal corresponding to the filtered signal to obtain a modified signal, and

quality measurement waveform in accordance with the modified ideal signal and the filtered signal.

39. The device according to § 38, in which the processor provides plenty of shifts by performing teams to ensure the shift in frequency, time shift and phase shift.

40. The device according to § 38, in which the processor compensates for the real signal through many changes by running the command

where y(t) - compensated real signal, x(t) is a real signal, t is time, j is the imaginary unit,the shift in frequency,- time shift,- phase shift.

41. The device according to § 38, in which the processor performs the filtering petamburan commands for assigning the compensated actual signal values, which is zero in the filtered intervals and zero in the other intervals.

42. The device according to paragraph 41, in which the processor performs the filtering by executing commands for assigning the compensated actual signal value that differs from zero for all elementary element of the real signal.

43. The device according to paragraph 41, in which the processor assigns compensated real signal some value by issuing commands to define a function with a value that is zero in intervals and filtered differs from zero in the other intervals, and multiplying the compensated actual signal to the specified function.

44. The device according to item 43, in which the processor determines the function by executing commands to define a function with values other than zero for all elementary element of the real signal.

45. The device according to § 38, in which the processor modifies the ideal signal by executing commands from the receipt of the modified ideal signal to obtain the value of zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals.

46. The device according to § 38, in which the processor modifies the ideal signal by executing commands to assign the ideal signal values of zero in intervals where f is lirovannye signal is set, zero and non-zero in the other intervals.

47. The device according to item 46, in which the processor assigns an ideal signal of a certain value by executing a command to define a function with a value of zero in intervals where the filtered signal has a value of zero and non-zero in the other intervals, and multiplying the compensated actual signal to the function.

48. The device according to § 42, in which the processor determines the quality of the waveform by performing the commands to calculate the first overall accuracy modulation.

49. The device according to p, in which the processor calculates the first modulation accuracy by performing teams to estimate the following equation:

where ρtotal-1first the overall accuracy of the modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and Rj,k=r[M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

50. The device according to p, in which the processor is additionally configured to, in the execution of commands to calculate a second total accuracy modulation.

51. The device according to item 50, in which the processor calculates the second modulation accuracy by performing teams to estimate the following equation:

where ρGeneral 2- second accuracy modulation, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element,- k-I readings in the j-th elementary element of the filtered signal, and- k-I readings in the j-th elementary ideal element signal.

52. The device according to paragraph 41, in which the processor determines the quality of the waveform by performing teams to determine the accuracy of modulation for the channel by time division.

53. The device according to paragraph 52, in which the processor calculates the accuracy of modulation for the channel by time division by performing teams to estimate the following equation:

where ρMrvan- accuracy modulation for channel Mrvan with time division, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k - index, hereafter the expectation of the increments in the elementary element, M - limit of summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and Rj,k=r[(M(j-1)+k] - k-I readings in the j-th elementary ideal element signal.

54. The device according to paragraph 41, in which the processor determines the parameters for measuring the quality of the signal by executing commands for calculating the coefficients of the power code domain.

55. The device according to item 54, in which the processor calculates the coefficients of the power code domain by executing the commands:

where ρMrvan,i- coefficient code domain channel Mrvan with time division and code channel i, w1the first code channel to channel Mrvan with time division, wvthe last code channel to channel Mrvan with time division, j is the index denoting the elementary element of the signal, N is the limit of the summation, indicating the number of elementary elements, k is the index denoting the increment in the elementary element, M is the limit of the summation, denoting the number of increments in the elementary element, Zj,k=z[(M(j-1)+k] - k-I readings in the j-th elementary element of the filtered signal, and R'i,j,k=R'i[(M(j-1)+k] - k-I readings in the j-th elementary element of the i-th code channel ID the real signal.



 

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