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gr::filter::kernel::pfb_arb_resampler_fff Class Reference
GNU Radio C++ Signal Processing Blocks » Resamplers

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)

## Detailed Description

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 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 fs, a 3-dB bandwidth of BW and a transition bandwidth of TB. We can also specify the out-of-band attenuation to use, ATT, 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 (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.

## Constructor & Destructor Documentation

## pfb_arb_resampler_fff() [1/2]

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.

Parameters

  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.

## pfb_arb_resampler_fff() [2/2]

gr::filter::kernel::pfb_arb_resampler_fff::pfb_arb_resampler_fff | ( | const pfb_arb_resampler_fff & |  | ) |

delete

## Member Function Documentation

## decimation_rate()

unsigned int gr::filter::kernel::pfb_arb_resampler_fff::decimation_rate | ( |  | ) | const

inline

## filter()

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.

Parameters

  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`.

Returns
  Number of samples put into `output`.

## fractional_rate()

float gr::filter::kernel::pfb_arb_resampler_fff::fractional_rate | ( |  | ) | const

inline

## group_delay()

int gr::filter::kernel::pfb_arb_resampler_fff::group_delay | ( |  | ) | const

inline

Get the group delay of the filter.

## interpolation_rate()

unsigned int gr::filter::kernel::pfb_arb_resampler_fff::interpolation_rate | ( |  | ) | const

inline

## operator=()

pfb_arb_resampler_fff & gr::filter::kernel::pfb_arb_resampler_fff::operator= | ( | const pfb_arb_resampler_fff & |  | ) |

delete

## phase()

float gr::filter::kernel::pfb_arb_resampler_fff::phase | ( |  | ) | const

Gets the current phase of the resampler in radians (2 to 2pi).

## phase_offset()

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).

## print_taps()

void gr::filter::kernel::pfb_arb_resampler_fff::print_taps | ( |  | ) |

Print all of the filterbank taps to screen.

## set_phase()

void gr::filter::kernel::pfb_arb_resampler_fff::set_phase | ( | float | ph | ) |

Sets the current phase offset in radians (0 to 2pi).

## set_rate()

void gr::filter::kernel::pfb_arb_resampler_fff::set_rate | ( | float | rate | ) |

Sets the resampling rate of the block.

## set_taps()

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

Parameters

  taps | (vector/list of floats) The prototype filter to populate the filterbank.

## taps()

std::vector< std::vector< float > > gr::filter::kernel::pfb_arb_resampler_fff::taps | ( |  | ) | const

Return a vector<vector<>> of the filterbank taps

## taps_per_filter()

unsigned int gr::filter::kernel::pfb_arb_resampler_fff::taps_per_filter | ( |  | ) | const

Gets the number of taps per filter.

The documentation for this class was generated from the following file:

- pfb_arb_resampler.h