Method for compensating gain flatness of transceiver
The present disclosure provides a method for compensating gain flatness of a transceiver including: a method for compensating gain flatness of a receiver, which compensates gain flatness of a receiving channel by using a complex-coefficient FIR filter in digital domain; and a method for compensating gain flatness of a transmitter, which compensates gain flatness of a transmitting channel by using a complex-coefficient FIR filter in digital domain. The method according to the present disclosure can balance compensation accuracy and calculation amount flexibly, and can focus on compensating the gain flatness at an edge of a frequency band, obtaining good performance with less calculation amount.
1. A method for compensating gain flatness of a receiver, by compensating a gain flatness of a receiving channel using a complex-coefficient finite impulse response (FIR) filter in digital domain, the method comprising:
converting a signal received by an analog-to-digital converter (ADC) into an IQ signal with zero-intermediate frequency;
setting f c −f s /2 as a starting frequency point, transmitting single-tone signals at N frequency points with a frequency interval of f s /N, and calculating a power P n of each frequency point in the digital domain, wherein f c represents a center frequency of the receiving channel, f s represents a sampling rate of a digital signal, and N takes a value of an integer power of 2;
calculating the gain flatness in a whole f s bandwidth based on a power of the center frequency f c of the receiving channel to obtain a sequence P n ′;
adding linear phase information to the sequence P′ n to construct a complex sequence X n of N points;
performing an inverse fast Fourier transform (IFFT) transformation of N points on the complex sequence X n to obtain a frequency response Y n ;
approximating the frequency response Y n using a complex sequence Z q of Q points, wherein Q is chosen from a suitable integer; and
constructing a Q-order complex FIR filter with the complex sequence Z n as a coefficient of the FIR filter in the digital domain, filtering the IQ signal, and obtaining a filtering result as data after compensating the gain flatness.
2. The method of claim 1 , wherein a power of an input signal is calculated by the following formula by using an amplitude of the IQ signal in the digital domain:
P= 10*log 10 ((Σ m=0 M−1 I m 2 +Q m 2 )/ M ),
wherein M represents the number of points for calculating the power.
3. The method of claim 1 , wherein the sequence P n ′ is calculated by the following formula:
P n ′=10 (P n −P N/2 )/20 , n= 0,1,2 . . . N− 1,
wherein P N/2 is a power point corresponding to the center frequency f c of the receiving channel.
4. The method of claim 1 , wherein the adding the linear phase information to the sequence P′ n to construct the complex sequence X n of N points is calculated by the following formula:
X n =P n ′*e jπ(N/2−n)*(N−1)/N , n= 0,1,2 . . . N− 1.
5. The method of claim 1 , wherein the performing the IFFT transformation of N points on the complex sequence X n to obtain the frequency response Y n comprising:
performing a shift processing on the complex sequence X n to obtain a complex sequence X′ n , and
performing the IFFT transformation on the complex sequence X′ n to obtain the frequency response Y n ;
wherein the shift processing is performed by the following formula:
X
n
′
=
{
X
n
+
N
/
2
if
n
<
N
/
2
X
n
-
N
/
2
if
n
≥
N
/
2
.
6. The method of claim 1 , further comprising:
giving single-tone signals at a plurality of frequency points (f 1 , f 2 , . . . , f n ) with different amplitudes (G 1 , G 2 , . . . , G n ), wherein the single-tone signals at an edge of a frequency band are given an amplitude of a relative large G value, and the single-tone signals at other part of the frequency band are given an amplitude of a relative small G value;
using the single-tone signals at the plurality of frequency points as an excitation source of the FIR filter in the digital domain;
calculating half of the frequency response Y n to obtain half of the complex sequence Z q ; and
obtaining the complex sequence Z q for best approximating the frequency response Y n when a mean square error is the smallest or an self-adaptation process converges.
7. The method of claim 1 , wherein the complex-coefficient FIR filter is constructed with four coefficient FIR filters.
8. A method for compensating gain flatness of a transmitter, by compensating a gain flatness of a transmitting channel using a complex-coefficient finite impulse response (FIR) filter in digital domain, the method comprising:
generating, by a numerically controlled oscillator (NCO), single-tone signals at N frequency points with a frequency interval of f s /N in a frequency range from −f s /2 to f s /2, and measuring the power P n at each frequency point, wherein f s represents a sampling rate of a digital signal, and N takes a value of an integer power of 2;
calculating the gain flatness of the transmitting channel based on a power of a center frequency f c of the transmitting channel to obtain a sequence P n ′;
adding linear phase information to the sequence P′ n to construct a complex sequence X n of N points;
performing an inverse fast Fourier transform (IFFT) transformation of N points on the complex sequence X n to obtain a frequency response Y n ;
approximating the frequency response Y n using a complex sequence Z q of Q points, wherein Q is chosen from a suitable integer; and
constructing a Q-order complex FIR filter with the complex sequence Z n as a coefficient of the FIR filter in the digital domain before the digital signal is transmitted to a digital-to-analog converter (DAC), converting a signal received by the ADC into an IQ signal, filtering the IQ signal, and obtaining a filtering result as transmitting data after compensating the gain flatness.
9. The method of claim 8 , wherein a power of an input signal is calculated by the following formula by using an amplitude of the IQ signal in the digital domain:
P
=
1
0
*
log
1
0
(
(
∑
m
=
0
M
-
1
I
m
2
+
Q
m
2
)
/
M
)
wherein M represents the number of points for calculating the power.
10. The method of claim 8 , wherein the sequence P n ′ is calculated by the following formula:
P n ′=10 (P n −P N/2 )/20 , n= 0,1,2 . . . N− 1,
wherein P N/2 is a power point corresponding to the center frequency f c of the transmitting channel.
11. The method of claim 8 , wherein the adding the linear phase information to the sequence P′ n to construct the complex sequence X n of N points is calculated by the following formula:
X n =P n ′*e jπ(N/2−n)*(N−1)/N , n= 0,1,2 . . . N− 1.
12. The method of claim 8 , wherein the performing the IFFT transformation of N points on the complex sequence X n to obtain the frequency response Y n comprising:
performing a shift processing on the complex sequence X n to obtain a complex sequence X′ n , and
performing the IFFT transformation on the complex sequence X′ n to obtain the frequency response Y n ;
wherein the shift processing is performed by the following formula:
X
n
′
=
{
X
n
+
N
/
2
if
n
<
N
/
2
X
n
-
N
/
2
if
n
≥
N
/
2
.
13. The method of claim 8 , further comprising:
giving single-tone signals at a plurality of frequency points (f 1 , f 2 , . . . , f n ) with different amplitudes (G 1 , G 2 , . . . , G n ), wherein the single-tone signals at an edge of a frequency band are given an amplitude of a relative large G value, and the single-tone signals at other part of the frequency band are given an amplitude of a relative small G value;
using the single-tone signals at the plurality of frequency points as an excitation source of the FIR filter in the digital domain;
calculating half of the frequency response Y n to obtain half of the complex sequence Z q ; and
obtaining the complex sequence Z q for best approximating the frequency response Y n when a mean square error is the smallest or an self-adaptation process converges.
14. The method of claim 8 , wherein the complex-coefficient FIR filter is constructed with four coefficient FIR filters.