A dynamic configurable audio re-acquisition method based on software radio

Abstract

The invention discloses a dynamically configurable audio re-collection method based on software radio, relates to the technical field of communication software radio, solves the problem that an IP core and a filter used in an audio re-collection module in the prior art occupy a large amount of resources, generates an NCO through a Cordic core, reloads coefficients in a filter module, and automatically cuts off the bit width, thereby saving a large amount of BRAM and LUT resources, and sends control commands through a controller written in an ARM to configure an FPGA module, thereby improving the efficiency of the system, realizing dynamic configuration, and complying with the idea of ​​software radio; on a reliable software radio platform, filter coefficients and bandwidth frequency configuration instructions are sent to an audio re-collection module through an ARM, thereby greatly saving resources on the FPGA and improving the flexibility of the project.

Description

Technical Field

[0001] The present invention relates to the technical field of communication software radio, and more specifically to the technical field of a dynamic configurable audio re-collection method based on software radio. Background Art

[0002] The voice signal receiver plays a very important role in electronic warfare. It can demodulate audio information without distortion, warn target signals, and obtain intelligence at the same time. Multi-sampling rate signal processing systems are widely used in voice processing, communications, radar and other fields. When audio demodulation of signals with different sampling rates is performed, multiple sampling rates can be achieved to make audio suitable for more scenarios.

[0003] The application block diagram of the audio re-collection module is as follows: Figure 1 As shown in the figure, taking the digital intermediate frequency receiver as an example, the antenna receives the signal, the analog signal is amplified to the appropriate amplitude through the RF module, the frequency is converted to the appropriate position, the analog intermediate frequency signal is converted into a digital intermediate frequency signal through the ADC, and then the low-speed baseband signal is output through digital down-conversion. The baseband signal is demodulated and sent to the audio re-sampling module for down-sampling, filtering, and re-sampling to obtain the corresponding audio sampling rate. The structural block diagram of the audio re-sampling module is shown in the figure. Figure 2 As shown in the figure, the module first converts the data coming from AD to baseband, thereby reducing the data rate, saving hardware resources and facilitating post-processing. Then, after demodulation, the audio signal can be restored and enter the audio filter module to filter out the audio signal of a specific bandwidth, and then the corresponding re-sampling module is selected to obtain the required sampling rate.

[0004] As one of the key technologies of software radio, the audio re-sampling module plays a key role in audio signal reception. Due to the improvement of receiver function requirements and the increase in the demand for variable bandwidth sampling rate, the IP core and filter used in the traditional audio re-sampling module will occupy a lot of resources. Summary of the invention

[0005] The purpose of the present invention is to solve the problem that the IP core and filter used in the audio re-collection module in the prior art will occupy a large amount of resources. In order to solve the above technical problems, the present invention provides a dynamically configurable audio re-collection method based on software radio.

[0006] In order to achieve the above-mentioned purpose, the present invention specifically adopts the following technical solutions:

[0007] A method for re-collecting dynamically configurable audio based on software radio comprises the following steps:

[0008] Step 1: Calculate the frequency control word PINC and the number of iterations i required for NCO mixing;

[0009] Step 2: Allocate the decimation factor of the filter according to the digital down-conversion bandwidth requirement;

[0010] Step 3: According to the requirements of the audio sampling output sampling rate and bandwidth, configure the decimation filter and interpolation filter in the audio downsampling;

[0011] Step 4: Generate filter coefficients using Matlab based on image suppression, output filter bandwidth, sampling rate, and rectangular coefficient requirements;

[0012] Step 5: Get the gain value of each filter and calculate the cutoff value;

[0013] Step 6: Set the filter in step 3 to coefficient reload mode;

[0014] Step 7: Take out the symmetrical half of the filter coefficients in step 4 and add the truncation value in step 5 to the end to form the coefficients sent in the coefficient overload mode, and store the coefficients in an array to be sent to ARM;

[0015] Step 8: In ARM, operate the controller register, first allocate different storage spaces for n groups of filter coefficients, and store them in BRAM;

[0016] Step 9: Send the coefficient reload instruction in ARM, and then send the corresponding coefficient from BRAM to the FPGA module command interface;

[0017] Step 10: In the filter module, the filter is truncated according to the truncation value in the filter coefficient sent down, thereby realizing the dynamic configuration of the audio re-sampling module.

[0018] Furthermore, the specific method of step 1 is:

[0019] Configure the NCO in the down-conversion circuit. The corresponding frequency signal can be obtained through the frequency control word. The mixer uses the Cordic algorithm. Assuming that the system clock is f clk , the input signal sampling rate is fs_in, the output sampling rate is fs_out, and the output signal frequency is f req , the bit width of the phase accumulator is N, and the value of the frequency control word is calculated according to formula (1);

[0020]

[0021] According to formula (2), the number of iterations i required for the Cordic algorithm is obtained:

[0022]

[0023] The calculation result is sent as a parameter to the digital down-conversion module.

[0024] Furthermore, the specific method of step 2 includes: according to the digital down-conversion output sampling rate requirement fs_out, obtaining the filter decimation multiple as fs_out / fs_in=D1.

[0025] Furthermore, the specific method of step 3 is:

[0026] According to the requirements of different sampling rates and different bandwidths after audio downsampling, assuming the input audio sampling rate is fs_audio_in and the output sampling rate is fs_audio_out, calculate the decimation multiple D from the input to the output end = fs_audio_out / fs_audio_in, and according to the decimation multiple D, allocate the decimation multiple of the filter. By coordinating the decimation multiples in steps 2 and 3, the output audio signal reaches an output sampling rate that is a non-integer multiple of the input sampling rate.

[0027] Furthermore, the specific method of step 5 is:

[0028] The filter generated in step 4 is fixed-pointed and passed Get the filter gain value; where B is the truncation bit width, a n are the filter coefficients and N is the filter order.

[0029] Furthermore, the specific method of step 8 is:

[0030] The AD data is used as the input signal for down-conversion. According to the Cordic algorithm, the signal is mixed and moved to the baseband, and the signal is automatically truncated using a barrel shifter.

[0031] Furthermore, the specific method of step 9 is:

[0032] In ARM, the filter coefficients in down-conversion and resampling are sent to each filter, the filter cutoff value is updated and sent to each module, and the allocation of the decimation multiples in the entire module is sent to the module to complete the selection of signal bandwidth. The filter decimation and interpolation multiples in the down-conversion and audio filtering modules and the audio up-sampling module are fixed.

[0033] The beneficial effects of the present invention are as follows:

[0034] (1) In the audio sampling module of the present invention, NCO is generated by Cordic core, and the coefficients in the filter module are reloaded, and the bit width is automatically truncated, which can save a lot of BRAM and LUT resources. Through the controller written in ARM, control commands are issued to configure the FPGA module, which improves the efficiency of the system and realizes dynamic configuration, which is in line with the idea of ​​software radio;

[0035] (2) The dynamically configurable audio re-sampling method based on software radio of the present invention, on a reliable software radio platform, sends filter coefficients and bandwidth frequency configuration instructions to the audio re-sampling module through ARM, which greatly saves resources on FPGA and improves the flexibility of the project. BRIEF DESCRIPTION OF THE DRAWINGS

[0036] Figure 1 This is the application block diagram of the audio resampling module in the receiver;

[0037] Figure 2 This is a specific implementation diagram of the structure of the audio re-collection module;

[0038] Figure 3 The present invention provides a structural block diagram of a dynamically configurable audio re-collection module. DETAILED DESCRIPTION

[0039] To make the purpose, technical solution and advantages of the embodiments of the present invention more clear, the technical solution in the embodiments of the present invention will be clearly and completely described below in conjunction with the accompanying drawings in the embodiments of the present invention.

[0040] Example 1

[0041] like Figure 3 As shown, the present invention provides a method for re-collecting dynamically configurable audio based on software radio, which mainly includes the following steps:

[0042] Step 1: Configure the NCO in the down-conversion circuit. The corresponding frequency signal can be obtained through the frequency control word. The mixer uses the Cordic algorithm. Assume that the system clock is f clk , the input signal sampling rate is fs_in, the output sampling rate is fs_out, and the output signal frequency is f req , the bit width of the phase accumulator is N, and the value of the frequency control word is calculated according to formula (1);

[0043]

[0044] According to formula (2), the number of iterations i required by the Cordic algorithm can be solved, and the calculation result is sent to the digital down-conversion module as a parameter;

[0045]

[0046] Step 2: According to the digital down-conversion output sampling rate requirement fs_out, the filter decimation factor is obtained as fs_out / fs_in=D1.

[0047] Step 3: According to the requirements of different sampling rates and bandwidths after audio downsampling; assuming that the input audio sampling rate is fs_audio_in and the output sampling rate is fs_audio_out, calculate the decimation multiple D from the input end to the output end = fs_audio_out / fs_audio_in; according to the decimation multiple D, allocate the decimation multiple of the filter; by coordinating the decimation multiples in steps 2 and 3, the output audio signal can reach an output sampling rate that is a non-integer multiple of the input sampling rate;

[0048] Step 4: According to the requirements of image suppression, output filter bandwidth and sampling rate, use the fdatool in Matlab to generate filters with different sampling rates and bandwidths; make the audio filter module filter remove the clutter after the audio signal is demodulated and downsampled, and the audio upsampling module filter completes the signal sampling rate increase;

[0049] Step 5: Fixed-point the filter generated in step 4 and pass Get the filter gain value; where B is the truncation bit width, an is the filter coefficient, and N is the filter order;

[0050] Step 6: Take out half of the coefficient generated in step 5 and add it to the truncated value to form a reload coefficient in preparation for ARM release;

[0051] Step 7: Repeat steps 4 to 6 to generate the interpolation filter in the audio resampling module, so that the audio signal is downsampled and filtered before being upsampled to reach the target sampling rate; and store the file in preparation for ARM to send it down;

[0052] Step 8: The AD data is used as the input signal for down-conversion. According to the Cordic algorithm, the signal is mixed and moved to the baseband. The barrel shifter is used to automatically truncate the signal.

[0053] Step 9: In ARM, filter coefficients in down-conversion and resampling are sent to each filter; filter cutoff values ​​are updated and sent to each module; and the allocation of decimation multiples in the entire module is sent to the module to complete the selection of signal bandwidth; the filter decimation and interpolation multiples in the down-conversion and audio filtering modules and the audio up-sampling module are fixed;

[0054] Step 10: In the filter module, the filter is truncated according to the truncation value in the filter coefficient sent down, thereby realizing the dynamic configuration of the audio re-sampling module.

Claims

1. A method for re-collecting dynamically configurable audio based on software radio, characterized in that: The following steps are involved: Step 1: Calculate the frequency control word PINC and the number of iterations i required for NCO mixing; Step 2: Allocate the decimation factor of the filter according to the digital down-conversion bandwidth requirements; Step 3: According to the requirements of the audio sampling output sampling rate and bandwidth, configure the decimation filter and interpolation filter in the audio downsampling; Step 4: Generate filter coefficients using Matlab based on image suppression, output filter bandwidth, sampling rate, and rectangular coefficient requirements; Step 5: Get the coefficients of each filter and calculate the cutoff value; Step 6: Set the filter in step 3 to coefficient reload mode; Step 7: Take out the symmetrical half of the filter coefficients in step 4 and add the truncation value in step 5 to the end to form the coefficients sent in the coefficient overload mode, and store the coefficients in an array to be sent to ARM; Step 8: In ARM, operate the controller register, first allocate different storage spaces for n groups of filter coefficients, and store them in BRAM; Step 9: Send the coefficient reload instruction in ARM, and then send the corresponding coefficient from BRAM to the FPGA module command interface; Step 10: In the filter module, the filter is truncated according to the truncation value in the filter coefficient sent down, thus realizing the dynamic configuration of the audio re-collection module; The specific method of step 1 is: Configure the NCO in the down-conversion circuit. The corresponding frequency signal can be obtained through the frequency control word. The mixer uses the Cordic algorithm. Assume that the system clock is f clk The input signal sampling rate is fs_in , output sampling rate fs_out , the output signal frequency is f req , the bit width of the phase accumulator is N , calculate the value of the frequency control word according to formula (1); （1）， According to formula (2), the number of iterations i required for the Cordic algorithm is obtained: （2）， Send the calculation result as a parameter to the digital down-conversion module; The specific method of step 5 is: The filter generated in step 4 is fixed-pointed and passed Get the filter truncation width; where B is the truncation width, a n are the filter coefficients and N is the filter order.

2. A method for re-collecting dynamically configurable audio based on software radio according to claim 1, characterized in that: The specific method of step 2 includes: according to the input signal sampling rate fs_in and digital down-conversion output sampling rate requirements fs_out , the filter decimation factor is fs_out / fs_in=D1 .

3. The method for re-collecting dynamically configurable audio based on software radio according to claim 1, characterized in that: The specific method of step 3 is: According to the different sampling rates and bandwidth requirements after audio downsampling, assuming that the input audio sampling rate fs_audio_in, Output sampling rate fs_audio_out, Calculate the decimation factor from input to output D = fs_audio_out / fs_audio_ in According to the decimation factor D, the decimation factor of the filter is allocated, and the decimation factors in step 2 and step 3 are coordinated to make the output audio signal reach an output sampling rate that is a non-integer multiple of the input sampling rate.

4. The method for re-collecting dynamically configurable audio based on software radio according to claim 1, characterized in that: The specific method of step 8 is: The AD data is used as the input signal for down-conversion. According to the Cordic algorithm, the signal is mixed and moved to the baseband, and the signal is automatically truncated using a barrel shifter.

5. The method for re-collecting dynamically configurable audio based on software radio according to claim 1, characterized in that: The specific method of step 9 is: In ARM, the filter coefficients in down-conversion and resampling are sent to each filter, the filter cutoff value is updated and sent to each module, and the allocation of the decimation multiples in the entire module is sent to the module to complete the selection of signal bandwidth. The filter decimation and interpolation multiples in the down-conversion and audio filtering modules and the audio up-sampling module are fixed.

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