CN101149630A - DDS signal source amplitude-frequency characteristic compensation method and related DDS signal source - Google Patents
DDS signal source amplitude-frequency characteristic compensation method and related DDS signal source Download PDFInfo
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Abstract
This invention offers a kind of amplitude frequency compensation method for the DDS signal source through the digital filter and the corresponding DDS signal source and adds a digital filter module and a selecting switch between the present DDS signal source's RAM search table and the digit and analog converter. Before the DDS signal source is used in reality set up a selecting switch and make the RAM finding table found the output data and then send the DAC directly and do the digital and analog convert. Then it can get the amplitude frequency characteristic function of signal source by testing the output signal and get the reciprocal of this function and then calculate it by using the frequency sampling method and then get the tapping coefficient of digital filter module. Write the tapping coefficient in the tapping coefficient finding table of digital filter module and set up the selecting switch again then let the output data of RAM finding table output to the DAC through the digital filter module and then do the digit and analog converter so it can realize the amplitude frequency characteristic correction for the DDS signal source.
Description
Technical Field
The invention relates to the field of DDS signal generators, in particular to a method and a device for effectively improving uneven amplitude-frequency characteristics of signals caused by DAC (digital-to-analog converter) holding characteristics and ripple fluctuation in a pass band of an elliptic filter.
Background
DDS (direct digital synthesis) is a third generation frequency synthesis technology following a direct frequency synthesis technology and a phase-locked loop frequency synthesis technology, and has been widely used in the field of signal generators in recent years by virtue of its characteristics of wide relative bandwidth, short conversion time, high resolution, continuous output phase, capability of generating wideband orthogonal signals and other various modulation signals, programmability, full digitalization, flexibility and convenience in control, and the like. The basic structure of the DDS signal source mainly comprises a phase accumulator, a waveform RAM, a digital-to-analog converter (DAC), a low-pass filter and the like. The phase accumulator is added with the phase increment determined by the frequency control word under the control of the sampling clock; the high-order output of the phase accumulator is used as the address of the waveform RAM to realize the conversion from the waveform phase to the amplitude; the waveform data is converted into analog quantity by DAC, and relatively smooth waveform is output by low-pass filter.
The low-pass filter plays roles of keeping effective components and inhibiting clutter in the DDS, and plays a vital role in the quality of output waveforms. For sinusoidal signals, harmonic components of the output signal will appear in kf according to nyquist sampling theory S ±f 0 Where f is S Is the frequency of the sampling clock, f 0 Is the frequency of the output signal. An elliptic filter with steep transition characteristics is usually selected from a DDS signal source to filter sinusoidal signals, so that harmonic suppression of the sinusoidal signals is ensured, but the amplitude-frequency characteristics of the elliptic filter have ripple fluctuation in a pass band and a stop band. In addition, as the current DAC realizes the ideal unit impact sampling, the zero-order hold technology is adopted, and the transfer function has the envelope characteristic of sinx/x. Therefore, the signal source based on the DDS has uneven passband amplitude-frequency characteristics.
At present, a DDS signal source usually adopts an LC resonance circuit and combines a software method to correct the amplitude-frequency characteristic, but due to the deviation of inductance and capacitance values and the influence of parasitic capacitance and parasitic inductance of a circuit board, the realized LC resonance circuit has a certain deviation with a theoretical calculation value, and the DDS signal source needs to be adjusted by professional designers every time, which wastes time and labor. The software correction method can only realize correction on dot frequency, and cannot realize amplitude correction of waveforms such as frequency sweeping and frequency modulation.
Disclosure of Invention
The invention aims to provide a technology for compensating amplitude-frequency characteristics of a DDS signal generator (also called a DDS signal source) through digital filtering.
The invention is characterized in that a digital filtering module and a selection switch (see figure 1) are additionally arranged between a RAM lookup table and a digital-to-analog converter (DAC) of the existing DDS signal source, amplitude information output by the RAM lookup table is switched by the selection switch, and whether the amplitude information is subjected to amplitude pre-correction by the digital filtering module and then subjected to digital-to-analog conversion or is directly sent to the DAC for digital-to-analog conversion is selected. Before the DDS signal source is put into practical application, amplitude-frequency characteristic compensation (namely correction) is carried out through digital filtering, and the specific correction steps are as follows:
1) Setting a selection switch, selecting to directly send data output by the RAM lookup table to a DAC (digital-to-analog converter) for digital-to-analog conversion:
2) Obtaining an amplitude-frequency characteristic function of a signal source by testing an output signal of the DDS signal source;
3) Taking the reciprocal of the obtained amplitude-frequency characteristic function;
4) Calculating a tap coefficient of the digital filtering module by using a frequency sampling method according to the reciprocal of the amplitude-frequency characteristic function;
5) Filling the tap coefficient into a tap coefficient lookup table of the digital filtering module;
6) And setting a selection switch, and outputting the data output by the RAM lookup table to the DAC after passing through the digital filtering module.
Another object of the present invention is to provide a DDS signal source for implementing the above method for compensating amplitude-frequency characteristics. The DDS signal source of the invention is mainly composed of a waveform data synthesis module, a digital filtering module, a selection switch, a DAC and signal conditioning module, a central processing module (CPU) and corresponding processing software, and the DDS signal source is divided into the following parts:
1) The waveform data synthesis module mainly comprises a phase accumulator, an RAM lookup table, a sampling clock and a corresponding logic control circuit, and is responsible for completing the function of generating high-speed waveform data with required frequency under the condition of sampling clock signal synchronization.
2) The digital filtering module is the core of the present invention, and as shown in fig. 2, it includes a data storage unit, a tap coefficient lookup table, a multiplier, an accumulator, and a state control logic unit: a sampling clock in the waveform data synthesis module is respectively connected with the state control logic unit and the accumulator; the state control logic unit outputs control signals to control the data storage unit, the tap coefficient lookup table and the accumulator; the output of the data storage unit and the output of the tap coefficient lookup table are sent to the multiplier to complete multiplication, the output of the multiplier is sent to the accumulator, and the accumulator outputs the corrected amplitude information.
3) The selection switch is simultaneously connected with the RAM lookup table and the accumulator of the digital filtering module, and the selection switch selects whether the amplitude information is sent to the DAC for digital-to-analog conversion after being subjected to amplitude pre-correction directly or through the digital filtering module.
4) The DAC and the signal conditioning module are responsible for completing digital-to-analog conversion of the amplitude information after pre-correction, filtering the analog signal and adjusting the amplitude and offset of the signal.
5) The central processing module mainly comprises a high-performance DSP (digital signal processor), is mainly used for carrying out overall control on the whole system, is mainly used for work such as waveform data transmission, phase accumulator control, digital filtering module parameter setting and the like, and controls the selector switch.
The working principle of the DDS signal source of the invention is shown in figure 1, according to the frequency control word set by the CPU, the phase accumulator generates the address signal under the control of the sampling clock to search the RAMAddressing by a look-up table, converting the phase information of the waveform into amplitude information by a RAM look-up table, wherein the frequency f of the output signal o And a frequency control word f REG Sampling clock frequency f clk And the number of bits N of the phase accumulator is:
the amplitude information output by the RAM lookup table is switched by a selection switch, and whether the amplitude information is subjected to digital-to-analog conversion after amplitude pre-correction by a digital filtering module or is directly sent to a DAC (digital-to-analog converter) for digital-to-analog conversion is selected.
In the digital filtering module (see fig. 2), the state control logic unit changes the state under the control of the sampling clock, and outputs control information to the data storage unit, the tap coefficient lookup table and the accumulator unit according to different current states. The data storage unit stores the amplitude information output by the RAM lookup table in a corresponding storage position under the control of the state logic control unit and selects corresponding amplitude information to output to the multiplier, the tap coefficient lookup table also transmits a corresponding tap coefficient to the multiplier under the control of the state logic control unit, and the tap coefficient is obtained by the correction method steps 1) -4). The multiplier multiplies the amplitude information by the tap coefficient and sends the result to the accumulator unit for accumulation. The transfer function of the digital filter module realized by the process
Wherein: h (e) jw ) Is the frequency response of the digital filter module, ω is the angular frequency, h (N) is the tap coefficient, and N is a positive integer representing the number of points intercepted. The whole amplitude-frequency pre-correction digital filtering module can be realized in an FPGA (field programmable logic array).
And sending the pre-corrected waveform data to a DAC to realize conversion from a digital signal to an analog signal. And the signal output by the DAC is processed by a signal conditioning channel to obtain final waveform output.
In the DDS signal source, the phase accumulator, the digital filtering module and the selection switch can be integrated in a high-performance FPGA for implementation.
The invention provides a fully digital amplitude-frequency characteristic correction function for a DDS-based signal source (a function generator or an arbitrary waveform generator). The realization cost is low, and the cost performance is very high. The user only needs to set the relevant parameters of the amplitude-frequency characteristic curve to be corrected, and the corresponding tap coefficient of the digital filter can be obtained. And modifying corresponding parameters according to the obtained tap coefficients, thus finishing corresponding amplitude-frequency characteristic correction. In addition, because the amplitude-frequency characteristic corrected by the invention is obtained by testing the output end of the signal source, the object of calibration is not only uneven in amplitude-frequency characteristic caused by the sampling and holding of the filter and the DAC, but also uneven in amplitude-frequency characteristic caused by the influence of component values in subsequent parts such as fine adjustment, attenuation and amplification of channel amplitude.
Drawings
Fig. 1 is an overall schematic block diagram of a DDS signal source of the present invention.
Fig. 2 is a functional block diagram of a digital filtering module.
Detailed Description
The invention is further illustrated by the following examples, without in any way limiting the scope of the invention, with reference to the accompanying drawings.
As shown in fig. 1, in the DDS signal source of the present invention, the waveform data synthesis module mainly includes a high-speed phase accumulator, a RAM lookup table, and a sampling clock. The RAM lookup table may be selected according to the waveform storage depth and sampling rate to be achieved by the designed signal source, and the storage depth is generally selected to be 64k × 12bit. The high-speed phase accumulator is also designed according to the requirement of the frequency resolution of a signal source, and the formula delta f = f is met clk /2 N Wherein f is clk When samplingThe clock frequency, N the number of bits of the phase accumulator, and Δ f the frequency resolution. At present, the FPGA has high working speed and rich internal resources, and is the first choice of high-speed phase accumulation design.
The acquisition of the DDS signal source amplitude-frequency characteristic function is mainly completed through a vector network analyzer. The vector network analyzer is actually a sweep source and is additionally provided with a spectrum analyzer, and the amplitude-frequency characteristic of a channel of the designed signal generator can be obtained through the vector network analyzer.
The digital filtering module, as shown in fig. 2, includes a data storage unit, a tap coefficient lookup table, a high-speed multiplier, a high-speed accumulator, and a state control logic unit. Tap coefficients h (0) to h (N-1) can be calculated according to the obtained amplitude-frequency characteristics of the signal generator, and the calculation method adopts a frequency sampling method. And filling the calculated tap coefficients into a tap coefficient lookup table of the digital filtering module. The number of tap coefficients depends on the operating frequency and error requirement of the system, and N is more than 16. The implementation of digital filtering is also fully integrated in the FPGA.
The CPU recommends the use of a high-speed, high-performance embedded processor for scheduling the entire system, including controlling the frequency control words of the phase accumulator, transmitting the waveform data of the RAM lookup table, setting the parameters of the digital filtering module, and controlling the selector switch.
The invention is applied to a DDS signal source of a CPCI interface, the sampling frequency of the signal source is 40MSa/S, and the maximum output sinusoidal signal frequency is 15MHz. The amplitude comparison table before and after calibration was shown in table 1 when the signal source output amplitude was measured to be 500mV using a hand-held spectrometer FSH3 from rodschwatz. It can be seen that the maximum difference of the amplitude unevenness before the correction is 2.5dB, and the difference is reduced to 0.5dB after the correction. The amplitude-frequency characteristic of the signal source is well improved.
TABLE 1
| Output frequency | Amplitude before correction (dBmV) | Corrected amplitude (dBmV) |
| 100kHz 500KHz 1MHz 5MHz 10MHz 15MHz | 54.0 53.9 53.8 53.4 52.5 51.5 | 54.0 54.0 53.9 53.9 53.6 54.1 |
Claims (7)
1. A DDS signal source amplitude-frequency characteristic compensation method is characterized in that a digital filtering module and a selection switch are arranged between an RAM lookup table and a digital-to-analog converter of a DDS signal source, and amplitude-frequency characteristic correction is carried out according to the following steps:
1) Setting a selection switch, and selecting to directly send data output by the RAM lookup table to a digital-to-analog converter for digital-to-analog conversion;
2) Obtaining an amplitude-frequency characteristic function of a signal source by testing an output signal of the DDS signal source;
3) Taking the reciprocal of the obtained amplitude-frequency characteristic function;
4) Calculating a tap coefficient of the digital filtering module by using a frequency sampling method according to the reciprocal of the amplitude-frequency characteristic function;
5) Filling the tap coefficient into a tap coefficient lookup table of the digital filtering module;
6) And a selection switch is arranged, so that the data output by the RAM lookup table is output to the digital-to-analog converter after passing through the digital filtering module.
2. The DDS signal source amplitude-frequency characteristic compensation method as claimed in claim 1, wherein: and 2) acquiring an amplitude-frequency characteristic function of the DDS signal source by using a vector network analyzer.
3. A DDS signal source comprising:
a central processing module for overall control of the system;
a waveform data synthesis module for generating waveform data with required frequency;
a digital-to-analog converter that converts waveform data from a digital signal to an analog signal; and
the signal conditioning module is used for filtering the analog signal, adjusting the amplitude and the offset of the signal and outputting the signal;
the DDS signal source is characterized in that a digital filtering module and a selection switch are further arranged between the waveform data synthesis module and the analog-to-digital converter, the digital filtering module carries out amplitude pre-correction on the waveform data amplitude information output by the waveform data synthesis module and then sends the waveform data amplitude information to the selection switch, the selection switch is simultaneously connected with the waveform data synthesis module and the digital filtering module and switches the waveform data amplitude information under the control of the central processing module, and the selection is to send the waveform data amplitude information generated by the waveform data synthesis module to the digital-to-analog converter for digital-to-analog conversion after the amplitude pre-correction is carried out directly or through the digital filtering module.
4. The DDS signal source of claim 3 wherein: the waveform data synthesis module mainly comprises a phase accumulator, an RAM lookup table, a sampling clock and a corresponding logic control circuit, wherein the phase accumulator generates an address signal to address the RAM lookup table under the control of the sampling clock according to a frequency control word set by the central processing module, and the RAM lookup table converts the phase information of the waveform into amplitude information.
5. The DDS signal source of claim 4, wherein said digital filter module comprises a data storage unit, a tap coefficient look-up table, a multiplier, an accumulator, and a state control logic unit, wherein: the state control logic unit and the accumulator are respectively connected with a sampling clock in the waveform data synthesis module; the data storage unit is connected with the output end of the RAM lookup table; the state control logic unit changes the state under the control of the sampling clock, and outputs control information to the data storage unit, the tap coefficient lookup table and the accumulator according to different current states, the data storage unit stores the amplitude information generated by the waveform data synthesis module in a corresponding storage position and selects corresponding amplitude information to output to the multiplier, the tap coefficient lookup table transmits the corresponding tap coefficient to the multiplier, the multiplier completes the multiplication of the amplitude information and the tap coefficient, and then sends the result to the accumulator unit for accumulation operation, and the corrected amplitude information is output.
6. The DDS signal source of claim 5, wherein said phase accumulator, digital filter module, and selection switch are integrated into a field programmable logic array.
7. The DDS signal source of any of claims 3-6, wherein said central processing module is an embedded digital signal processor.
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