3.1.1. Analysis and Verification of Nonlinear Characteristics of Signal Source
There is always a carrier frequency in the radar emitter. Because any carrier frequency is not absolutely stable the actual radar emitter carrier frequency will not be completely equal to its nominal frequency value, and there is always more or less deviation. The phase noise and spurious output of frequency source of transmitting equipment are one of the main sources of spurious components of transmitting signal. The phase noise and spurious output of different radar emitters lead to different spurious components of different individual signals. There are many literatures on the analysis of phase noise and spurious output of crystal oscillator and PLL frequency source. Considering that direct digital synthesizer (DDS) has become the mainstream of modern radar emitter frequency synthesizer, this section mainly analyzes the phase noise, harmonic and spurious components of DDS output signal. Next, the spectrum structure, phase noise and spectrum spurious of the ideal DDS output signal are studied, and various factors causing the spurious of DDS output signal, such as phase truncation spurious and amplitude quantization spurious, are comprehensively analyzed.
Figure 5 shows the schematic diagram of DDS, the accumulator outputs linear increasing phase value, which is used as address to address sine function table. The output of ROM is converted into analog signal by DAC. In
Figure 5, the word length of accumulator is L, the value of frequency control word is
, W is the number of ROM address lines, D is the word length of ROM data line and D/A converter. The output frequency formula of DDS is:
There are two main spurious sources in actual DDS system: phase truncation error, and amplitude quantization error. These two spurious sources are discussed below.
I. Phase truncation error.
According to
Figure 5, it can be seen that the number of phase truncation bits
. If only the spurious component caused by truncation is considered, only needs to analyze the output signal after phase to amplitude conversion. Assuming that there is no phase truncation, the discrete value of the output signal amplitude after addressing is:
After B-bit truncation, the discrete value of output signal is Equation (2), where
is the downward integer of
.
Assuming that the phase rounding error is
, the above equation can be converted as follows, where
means
to
remainder operation.
The error sequence is analyzed as follows: since
does not always return to zero after each overflow, it is assumed that the period of
is
, then
Suppose
is the greatest common divisor of
and
, then
It can be seen that the least common multiple of
and
is
. Then
, so the period of the error sequence
is Equation (8), where formula
means the greatest common divisor between
and
.
According to the theory of discrete digital signal, the frequency spectrum of repeats with the clock frequency of , then in the interval of , the frequency spectrum is composed of discrete spectral lines.
If the continuous form of the corresponding error after truncation is written as
, then
can be considered as the result of sampling
by clock cycle. It is easy to know that
is a sawtooth wave with amplitude of
and period of
. In order to analyze the spectrum characteristics of
, we need to expand it by Fourier series, but because its value at the discontinuous point is 0, it does not meet the Dirichlet condition, so it cannot be expanded directly. In order to solve this problem, Nicholas and Samueli propose a correction method: divide
into two parts
and
which satisfy the expansion condition, as shown in
Figure 6, the period of
is
and the pulse width is
, where:
After the correction, the Fourier expansion of the above two equations can be obtained:
Omit the DC component in the above formula, where
. The expansion is sampled as follows:
After detailed mathematical analysis of
, it can be expressed as the discrete weighted sum of
spectral lines as follows:
where
,
represent the amplitude and phase values of the
k-th corresponding frequency component of the error vector respectively [
32].
According to
, the output signal sequence caused by phase truncation is:
It can be seen that the error sequence of output signal caused by phase truncation can be written as:
By substituting
and the Fourier series expansion of
, we can get:
It can be seen from the above formula that the spurious component of the output signal is also composed of
spectral lines. By mapping them to the actual frequency range of
, the frequency value
and corresponding amplitude value
of each spectral line can be obtained as follows:
By further simplifying
, we can get:
It is easy to see from the above formula that when k = 1, the corresponding frequency and amplitude values of the maximum stray frequency component are as follows:
It can be further calculated that the SNR of single frequency output is:
Since
, we can infer that:
It can be seen that the spurious spectral line position of the output signal caused by the phase truncation of the direct digital waveform synthesis method is determined by the input frequency control word and the word length of the phase accumulator , and the spurious level is determined by the value of .
Based on the phase truncation of DDS system, the simulation of single frequency signal can verify that the spurious characteristics meet the above conclusions. The simulation parameters are set as shown in
Table 1, and the simulation results are shown in
Figure 7.
According to the above analysis of the phase truncation spurious component, substituting the parameters into Equations (20) and (21), and can be calculated. It can be seen that the simulation results are basically consistent with the theoretical derivation.
According to the digital characteristics of the direct digital frequency synthesis method, the frequency of the LFM signal generated by the direct digital frequency synthesis method is increasing step by step, so all the frequency components of the signal in the whole frequency band can be regarded as the superposition of the spurious components of each single frequency component. Since the frequency band range of DDS output signal is within
at most, the maximum spurious frequency corresponding to each frequency component in this range can be calculated one by one. As the output frequency increases, the maximum spurious frequency changes, as shown in
Figure 8. The parameters are set as L = 24, B = 18.
It can be seen from
Figure 8 that as the frequency of the output signal increases linearly with time, the corresponding maximum stray frequency also changes linearly with time, but the change rate is far greater than the frequency modulation slope of the signal, showing an irregular sawtooth wave. Calculate the frequency modulation slope
and
of signal and spurious respectively, define
to represent the change rate relationship between signal and spurious, it can be seen that when L-B = 6, the change slope of spurious is 65 times of that of signal, which is much larger than that of signal. Through simulation, it can be found that
changes with the change of L-B, and the overall change trend is shown in
Figure 9.
It is not difficult to see that with the increase of L-B, shows an exponential growth trend. Therefore, when the L-B value is large, the spurious bandwidth will be far greater than the signal bandwidth, and the larger the L-B value is, the more uniform the spurious distribution is. After filtering, the spurious distribution in the band is approximately uniform relative to the signal.
In a word, it can be considered that the spurious of DDS signal source due to phase truncation mainly reduces the signal-to-noise ratio of LFM signal source. Based on DDS phase truncation, the LFM signal is simulated, and the simulation parameters are set as shown in
Table 2.
From Equation (23), the signal-to-noise ratio of the output signal is
. The spectrum of the output signal is shown in
Figure 10. From the simulation results, it can be seen that the spurious component is reflected in the form of noise, and the simulation results are basically consistent with the theory.
II. Amplitude quantization error.
Since amplitude quantization occurs after phase truncation, the distribution characteristics of spurious signals caused by amplitude quantization are discussed in two cases: with and without phase truncation. Here we define ( and are prime to each other), the two cases are discussed as follows:
When
and there is no phase truncation, the spurious caused by quantization error can be expressed as follows:
It can be seen that
is a periodic sequence with a period of
, so the output spurious signal has at most
frequency components in the
interval. Therefore, the kth frequency value of the output spurious signal should be:
It can be proved that when k is even, the amplitude at frequency
is 0, so the spurious signal only contains the odd multiple frequency harmonic of
. The simulation parameter settings are shown in
Table 3.
Here
,
. It can be calculated that the frequency value of the output signal
, and the corresponding quantization spurious is
. The simulation results are shown in
Figure 11. It can be seen that the position and distribution of stray components are consistent with the theory.
When
, the phase truncation error occurs in the front end. According to Equation (3), the spurious component generated by quantization error can be expressed as:
According to the previous formula:
It is not difficult to see that the longest period of can be consistent with that of , which is . At this time, the period of amplitude quantization spurious signal is longer, and the corresponding distribution in frequency domain is more dispersed. When is odd, the period of can reach . If points DFT is performed on the signal, the spurious noise will be distributed on each spectral line, which can be considered as approximately uniformly distributed noise.
Considering the influence of amplitude quantization in the above two cases, it can be seen that when the LFM signal is output, because the frequency component of the output signal will occupy a certain bandwidth, the amplitude quantization spurious distribution corresponding to each different frequency value is also different.
Next, we analyze the noise power value caused by amplitude quantization, the amplitude quantization error is regarded as a random variable obeying uniform distribution in the range of
, where
and
is the quantization number.
is the sine function before quantization,
is the sine function after quantization, and
is the amplitude quantization error function. After digital to analog conversion, the amplitude quantization error function becomes a step waveform. Considering its periodicity, the time domain expression of its signal in one cycle is obtained as follows:
where
is the unit step function. The average power of
can be calculated as:
Here,
is regarded as a uniform distribution in the range of
, so it is more convenient to calculate its average power in a statistical sense as follows:
Assuming that the amplitude of the output signal is 1, the total SNR of the output signal can be expressed as:
Therefore, the total SNR of output signal amplitude quantization noise is only related to quantization level bits. Considering the phase truncation effect and amplitude quantization effect, the output LFM signal is simulated, and the parameter settings are shown in
Table 4.
According to the previous theoretical analysis, the amplitude quantization of the phase truncated signal is carried out. When the quantization bit is 8, the spectrum and amplitude quantization noise of the output LFM signal are shown in
Figure 12. The SNR caused by the amplitude quantization is about 50 dB, which is consistent with the theoretical calculation results.
3.1.2. Analysis and Verification of Nonlinear Characteristics of Mixer and Amplifier
The mixer and amplifier are the core modules in the transmission channel. The mixer is responsible for moving the IF signal spectrum to the RF region, and the amplifier is responsible for amplifying the input RF signal. However, the nonlinear characteristic models of the two are consistent, as shown in
Figure 13. The nonlinear characteristic parameters mainly include 1 dB compression point, third-order truncation point and second-order truncation point [
33,
34,
35]. The nonlinear effects are mainly high-order harmonics and intermodulation distortion. Next, we analyze them respectively.
Assuming that the fourth-order and higher-order harmonics generated by nonlinearity can be ignored, the input nonlinearity can be expressed by a third-order power series. The relationship between input and output is as follows:
Among them,
and
respectively represent the signal amplitude at the input and the output. When the input signal is a single frequency signal:
The output signal can be expressed as:
It can be seen from the above formula that when the input is a single frequency signal, under the influence of nonlinear effect, compared with the power spectrum of the input signal, it is found that the power at different frequency points on the fundamental and adjacent channels is greater than that of the input signal, which is called spectrum growth [
36]. The output signal will produce the second and third harmonic of the fundamental frequency signal in addition to the fundamental frequency signal. The relationship between the coefficients of each component can be shown in
Table 5.
When the input is a dual frequency signal, the output signal will be generated according to the following rules:
It can be seen that the output signal not only has the required main signal components, but also produces DC components, second-order and third-order harmonics and the corresponding intermodulation components of second-order and third-order. The coefficients of each component are summarized in
Table 6.
Then, the nonlinear characteristics of 1 dB compression point, third-order truncation point and second-order truncation point are analyzed and verified.
I. 1 dB compression point
Due to the nonlinearity of the input channel, when the input signal power reaches
, the power gain of the output signal relative to the input signal will decrease 1 dB. The calculation of 1 dB compression point is deduced as follows: under the assumption of linear condition, the signal before and after entering the analog device has the following relationship:
That is, the amplitude of the fundamental frequency component of the output signal is
. When there is nonlinearity, according to the above analysis, the amplitude of the output fundamental frequency component should be
. According to the definition of
, it can be calculated by
. Let
, the power of the input signal can be expressed as:
This shows that when 1 dB gain compression occurs in the input, the amplitude of the received sinusoidal waveform will be , and the corresponding power value will be . It can be seen that the 1 dB compression point of the system is due to the increasing proportion of the harmonic component at the output end of the system and the corresponding reduction of the energy of the fundamental frequency component, resulting in the gain compression phenomenon.
II. Third order truncation point
When the dual frequency signal is input, the power of the output third-order intermodulation signal caused by nonlinearity is equal to the power of the output fundamental frequency signal, and the corresponding input signal power level is
. The following is a quantitative derivation of the expression of the third-order truncation point: Assuming that the corresponding waveform amplitude is
when the input reaches the third-order truncation point, according to the definition of
and the corresponding coefficient of the third-order intermodulation component obtained from the above analysis, we can get the following results:
The power of the input signal can be expressed as:
It can be seen that the value of the third-order truncation point mainly depends on the gain coefficient of the output fundamental frequency component and the gain coefficient of the third-order harmonic, the quantitative relationship between this index and 1 dB compression point can be calculated as follows:
Taking logarithm of the above formula, we can get the following results:
Furthermore, the relationship between the input signal power level and the third-order intermodulation component, and the third-order harmonic component level in the linear range can be calculated. When the power level of the input signal is
, the conversion gain is G, and the amplitude of the input signal is
, the amplitude of the corresponding fundamental frequency can be approximately considered as
. At this time, the amplitude of the third-order intermodulation component generated in the output signal is:
When converted to dBm, the following results can be obtained:
The suppression degree of the third-order intermodulation component relative to the fundamental frequency component at the corresponding output is as follows:
According to the analysis of the third-order harmonic component, the corresponding output third-order harmonic level is
, so the third-order harmonic level is:
The output third-order harmonic suppression is:
III. Second order truncation point
When the dual frequency signal is input, the power of the output second-order intermodulation signal caused by nonlinearity is equal to the power of the output fundamental frequency signal, and the corresponding input signal power level is
. The following is a quantitative derivation of the expression of the second-order truncation point: Assuming that the corresponding waveform amplitude is
when the input reaches the second-order truncation point, according to the definition of
and the corresponding coefficient of the second-order intermodulation component obtained from the above analysis, we can get the following results:
The power of the input signal can be expressed as:
Furthermore, the relationship between the input signal power level and the second-order intermodulation component, and the second-order harmonic component level in the linear range can be calculated. When the power level of the input signal is
, the conversion gain is G, and the amplitude of the input signal is
, the amplitude of the corresponding fundamental frequency can be approximately considered as
. At this time, the amplitude of the second-order intermodulation component generated in the output signal is:
When converted to dBm, the following results can be obtained:
The suppression degree of the second-order intermodulation component relative to the fundamental frequency component at the corresponding output is as follows:
According to the analysis of the second-order harmonic component, the corresponding output second-order harmonic level is
, so the second-order harmonic level is:
The output second-order harmonic suppression is:
According to the analysis of DDS in the previous paper, the IF output signal after ideal orthogonal modulation only contains useful signal components and noises of various components, as shown in
Figure 14. After mixing, amplifying and other processing, the RF output signal component should include the LO, useful signal and all order harmonics of the useful signal—mainly, the second-order and third-order harmonics. If the IF signal is expressed as:
Then according to the nonlinear characteristic model of mixer and amplifier, the RF signal can be expressed as:
The nonlinear parameters of the mixer and amplifier are set as shown in
Table 7. According to the theoretical calculation, the suppression of the second-order harmonic component relative to the fundamental frequency in the mixer output signal should be 35 dBm, and the suppression of the third-order harmonic component relative to the fundamental frequency should be 44 dBm. The output signal of mixer is shown in
Figure 15. It can be seen that the simulation results are basically consistent with the theory.
According to the theoretical calculation, the suppression of the third-order harmonic component relative to the fundamental frequency in the amplifier output signal should be 96 dBm. The output signal of amplifier is shown in
Figure 16. It can be seen that the simulation results are basically consistent with the theory.