Interpolating MMSE filter with complex input, complex output.
Constructor Specific Documentation:
Build the interpolating MMSE filter (complex input, complex output)
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declare_sample_delay(fractional_interpolator_cc_sptr self, unsigned int delay)
pc_input_buffers_full_avg(fractional_interpolator_cc_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(fractional_interpolator_cc_sptr self) -> pmt_vector_float
Interpolating MMSE filter with float input, float output.
Constructor Specific Documentation:
Build the interpolating MMSE filter (float input, float output)
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declare_sample_delay(fractional_interpolator_ff_sptr self, unsigned int delay)
pc_input_buffers_full_avg(fractional_interpolator_ff_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(fractional_interpolator_ff_sptr self) -> pmt_vector_float
resampling MMSE filter with complex input, complex output
Constructor Specific Documentation:
Build the resampling MMSE filter (complex input, complex output)
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declare_sample_delay(fractional_resampler_cc_sptr self, unsigned int delay)
pc_input_buffers_full_avg(fractional_resampler_cc_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(fractional_resampler_cc_sptr self) -> pmt_vector_float
Resampling MMSE filter with float input, float output.
Constructor Specific Documentation:
Build the resampling MMSE filter (float input, float output)
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declare_sample_delay(fractional_resampler_ff_sptr self, unsigned int delay)
pc_input_buffers_full_avg(fractional_resampler_ff_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(fractional_resampler_ff_sptr self) -> pmt_vector_float
Polyphase filterbank arbitrary resampler with gr_complex input, gr_complex output and float taps.
This block takes in a signal stream and calls gr::filter::kernel::pfb_arb_resampler_ccf to perform arbitrary resampling on the stream.
Output sampling rate is * input rate.
Constructor Specific Documentation:
Build the polyphase filterbank arbitray resampler.
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Gets the decimation rate of the filter.
declare_sample_delay(pfb_arb_resampler_ccf_sptr self, unsigned int delay)
Gets the fractional rate of the filter.
Get the group delay of the filter.
Gets the interpolation rate of the filter.
pc_input_buffers_full_avg(pfb_arb_resampler_ccf_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(pfb_arb_resampler_ccf_sptr self) -> pmt_vector_float
Gets the current phase of the resampler in radians (2 to 2pi).
Calculates the phase offset expected by a sine wave of frequency and sampling rate (assuming input sine wave has 0 degree phase).
Print all of the filterbank taps to screen.
Sets the current phase offset in radians (0 to 2pi).
Sets the resampling rate of the block.
Resets the filterbank’s filter taps with the new prototype filter
Return a vector<vector<>> of the filterbank taps
Gets the number of taps per filter.
Polyphase filterbank arbitrary resampler with float input, float output and float taps.
This block takes in a signal stream and performs arbitrary resampling. The resampling rate can be any real number . The resampling is done by constructing filters where is the interpolation rate. We then calculate where .
Using and , we can perform rational resampling where is a rational number close to the input rate where we have filters and we cycle through them as a polyphase filterbank with a stride of so that .
To get the arbitrary rate, we want to interpolate between two points. For each value out, we take an output from the current filter, , and the next filter and then linearly interpolate between the two based on the real resampling rate we want.
The linear interpolation only provides us with an approximation to the real sampling rate specified. The error is a quantization error between the two filters we used as our interpolation points. To this end, the number of filters, , used determines the quantization error; the larger , the smaller the noise. You can design for a specified noise floor by setting the filter size (parameters ). The size defaults to 32 filters, which is about as good as most implementations need.
The trick with designing this filter is in how to specify the taps of the prototype filter. Like the PFB interpolator, the taps are specified using the interpolated filter rate. In this case, that rate is the input sample rate multiplied by the number of filters in the filterbank, which is also the interpolation rate. All other values should be relative to this rate.
For example, for a 32-filter arbitrary resampler and using the GNU Radio’s firdes utility to build the filter, we build a low-pass filter with a sampling rate of , a 3-dB bandwidth of and a transition bandwidth of . We can also specify the out-of-band attenuation to use, , and the filter window function (a Blackman-harris window in this case). The first input is the gain of the filter, which we specify here as the interpolation rate ().
The theory behind this block can be found in Chapter 7.5 of the following book.
Constructor Specific Documentation:
Build the polyphase filterbank arbitray resampler.
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Gets the decimation rate of the filter.
declare_sample_delay(pfb_arb_resampler_fff_sptr self, unsigned int delay)
Gets the fractional rate of the filter.
Get the group delay of the filter.
Gets the interpolation rate of the filter.
pc_input_buffers_full_avg(pfb_arb_resampler_fff_sptr self) -> pmt_vector_float
pc_output_buffers_full_avg(pfb_arb_resampler_fff_sptr self) -> pmt_vector_float
Gets the current phase of the resampler in radians (2 to 2pi).
Calculates the phase offset expected by a sine wave of frequency and sampling rate (assuming input sine wave has 0 degree phase).
Print all of the filterbank taps to screen.
Sets the current phase offset in radians (0 to 2pi).
Sets the resampling rate of the block.
Resets the filterbank’s filter taps with the new prototype filter
Return a vector<vector<>> of the filterbank taps
Gets the number of taps per filter.