Polyphase filterbank arbitrary resampler with float input, float output and float taps. More...
#include <gnuradio/filter/pfb_arb_resampler.h>
Public Member Functions  
pfb_arb_resampler_fff (float rate, const std::vector< float > &taps, unsigned int filter_size)  
pfb_arb_resampler_fff (const pfb_arb_resampler_fff &)=delete  
pfb_arb_resampler_fff &  operator= (const pfb_arb_resampler_fff &)=delete 
void  set_taps (const std::vector< float > &taps) 
std::vector< std::vector< float > >  taps () const 
void  print_taps () 
void  set_rate (float rate) 
void  set_phase (float ph) 
float  phase () const 
unsigned int  taps_per_filter () const 
unsigned int  interpolation_rate () const 
unsigned int  decimation_rate () const 
float  fractional_rate () const 
int  group_delay () const 
float  phase_offset (float freq, float fs) 
int  filter (float *output, float *input, int n_to_read, int &n_read) 
Polyphase filterbank arbitrary resampler with float input, float output and float taps.
This takes in a signal stream and performs arbitrary resampling. The resampling rate can be any real number r. The resampling is done by constructing N filters where N is the interpolation rate. We then calculate D where D = floor(N/r).
Using N and D, we can perform rational resampling where N/D is a rational number close to the input rate r where we have N filters and we cycle through them as a polyphase filterbank with a stride of D so that i+1 = (i + D) % N.
To get the arbitrary rate, we want to interpolate between two points. For each value out, we take an output from the current filter, i, and the next filter i+1 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, N, used determines the quantization error; the larger N, the smaller the noise. You can design for a specified noise floor by setting the filter size (parameters filter_size). 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 32filter arbitrary resampler and using the GNU Radio's firdes utility to build the filter, we build a lowpass filter with a sampling rate of fs, a 3dB bandwidth of BW and a transition bandwidth of TB. We can also specify the outofband attenuation to use, ATT, and the filter window function (a Blackmanharris window in this case). The first input is the gain of the filter, which we specify here as the interpolation rate (32).
self._taps = filter.firdes.low_pass_2(32, 32*fs, BW, TB, attenuation_dB=ATT, window=fft.window.WIN_BLACKMAN_hARRIS)
The theory behind this block can be found in Chapter 7.5 of the following book:
f. harris, "Multirate Signal Processing for Communication Systems", Upper Saddle River, NJ: Prentice Hall, Inc. 2004.
gr::filter::kernel::pfb_arb_resampler_fff::pfb_arb_resampler_fff  (  float  rate, 
const std::vector< float > &  taps,  
unsigned int  filter_size  
) 
Creates a kernel to perform arbitrary resampling on a set of samples.
rate  (float) Specifies the resampling rate to use 
taps  (vector/list of floats) The prototype filter to populate the filterbank. The taps * should be generated at the filter_size sampling rate. 
filter_size  (unsigned int) The number of filters in the filter bank. This is directly related to quantization noise introduced during the resampling. Defaults to 32 filters. 

delete 

inline 
int gr::filter::kernel::pfb_arb_resampler_fff::filter  (  float *  output, 
float *  input,  
int  n_to_read,  
int &  n_read  
) 
Performs the filter operation that resamples the signal.
This block takes in a stream of samples and outputs a resampled and filtered stream. This block should be called such that the output has rate
* n_to_read
amount of space available in the output
buffer.
output  The output samples at the new sample rate. 
input  An input vector of samples to be resampled 
n_to_read  Number of samples to read from input . 
n_read  (out) Number of samples actually read from input . 
output
.

inline 

inline 
Get the group delay of the filter.

inline 

delete 
float gr::filter::kernel::pfb_arb_resampler_fff::phase  (  )  const 
Gets the current phase of the resampler in radians (2 to 2pi).
float gr::filter::kernel::pfb_arb_resampler_fff::phase_offset  (  float  freq, 
float  fs  
) 
Calculates the phase offset expected by a sine wave of frequency freq
and sampling rate fs
(assuming input sine wave has 0 degree phase).
void gr::filter::kernel::pfb_arb_resampler_fff::print_taps  (  ) 
Print all of the filterbank taps to screen.
void gr::filter::kernel::pfb_arb_resampler_fff::set_phase  (  float  ph  ) 
Sets the current phase offset in radians (0 to 2pi).
void gr::filter::kernel::pfb_arb_resampler_fff::set_rate  (  float  rate  ) 
Sets the resampling rate of the block.
void gr::filter::kernel::pfb_arb_resampler_fff::set_taps  (  const std::vector< float > &  taps  ) 
Resets the filterbank's filter taps with the new prototype filter
taps  (vector/list of floats) The prototype filter to populate the filterbank. 
std::vector<std::vector<float> > gr::filter::kernel::pfb_arb_resampler_fff::taps  (  )  const 
Return a vector<vector<>> of the filterbank taps
unsigned int gr::filter::kernel::pfb_arb_resampler_fff::taps_per_filter  (  )  const 
Gets the number of taps per filter.