GNU Radio Manual and C++ API Reference 3.7.14.0
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pfb_arb_resampler_fff.h
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1/* -*- c++ -*- */
2/*
3 * Copyright 2009-2012 Free Software Foundation, Inc.
4 *
5 * This file is part of GNU Radio
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22
23
24#ifndef INCLUDED_PFB_ARB_RESAMPLER_FFF_H
25#define INCLUDED_PFB_ARB_RESAMPLER_FFF_H
26
27#include <gnuradio/block.h>
28#include <gnuradio/filter/api.h>
29
30namespace gr {
31namespace filter {
32
33/*!
34 * \brief Polyphase filterbank arbitrary resampler with
35 * float input, float output and float taps
36 * \ingroup resamplers_blk
37 *
38 * \details
39 * This block takes in a signal stream and performs arbitrary
40 * resampling. The resampling rate can be any real number
41 * <EM>r</EM>. The resampling is done by constructing <EM>N</EM>
42 * filters where <EM>N</EM> is the interpolation rate. We then
43 * calculate <EM>D</EM> where <EM>D = floor(N/r)</EM>.
44 *
45 * Using <EM>N</EM> and <EM>D</EM>, we can perform rational
46 * resampling where <EM>N/D</EM> is a rational number close to the
47 * input rate <EM>r</EM> where we have <EM>N</EM> filters and we
48 * cycle through them as a polyphase filterbank with a stride of
49 * <EM>D</EM> so that <EM>i+1 = (i + D) % N</EM>.
50 *
51 * To get the arbitrary rate, we want to interpolate between two
52 * points. For each value out, we take an output from the current
53 * filter, <EM>i</EM>, and the next filter <EM>i+1</EM> and then
54 * linearly interpolate between the two based on the real
55 * resampling rate we want.
56 *
57 * The linear interpolation only provides us with an approximation
58 * to the real sampling rate specified. The error is a
59 * quantization error between the two filters we used as our
60 * interpolation points. To this end, the number of filters,
61 * <EM>N</EM>, used determines the quantization error; the larger
62 * <EM>N</EM>, the smaller the noise. You can design for a
63 * specified noise floor by setting the filter size (parameters
64 * <EM>filter_size</EM>). The size defaults to 32 filters, which
65 * is about as good as most implementations need.
66 *
67 * The trick with designing this filter is in how to specify the
68 * taps of the prototype filter. Like the PFB interpolator, the
69 * taps are specified using the interpolated filter rate. In this
70 * case, that rate is the input sample rate multiplied by the
71 * number of filters in the filterbank, which is also the
72 * interpolation rate. All other values should be relative to this
73 * rate.
74 *
75 * For example, for a 32-filter arbitrary resampler and using the
76 * GNU Radio's firdes utility to build the filter, we build a
77 * low-pass filter with a sampling rate of <EM>fs</EM>, a 3-dB
78 * bandwidth of <EM>BW</EM> and a transition bandwidth of
79 * <EM>TB</EM>. We can also specify the out-of-band attenuation to
80 * use, <EM>ATT</EM>, and the filter window function (a
81 * Blackman-harris window in this case). The first input is the
82 * gain of the filter, which we specify here as the interpolation
83 * rate (<EM>32</EM>).
84 *
85 * <B><EM>self._taps = filter.firdes.low_pass_2(32, 32*fs, BW, TB,
86 * attenuation_dB=ATT, window=filter.firdes.WIN_BLACKMAN_hARRIS)</EM></B>
87 *
88 * The theory behind this block can be found in Chapter 7.5 of the
89 * following book:
90 *
91 * <B><EM>f. harris, "Multirate Signal Processing for Communication
92 * Systems", Upper Saddle River, NJ: Prentice Hall, Inc. 2004.</EM></B>
93 */
94
96{
97public:
98 // gr::filter::pfb_arb_resampler_fff::sptr
99 typedef boost::shared_ptr<pfb_arb_resampler_fff> sptr;
100
101 /*!
102 * Build the polyphase filterbank arbitrary resampler.
103 * \param rate (float) Specifies the resampling rate to use
104 * \param taps (vector/list of floats) The prototype filter to populate the
105 *filterbank. The taps should be generated at the filter_size sampling rate. \param
106 *filter_size (unsigned int) The number of filters in the filter bank. This is
107 *directly related to quantization noise introduced during the resampling. Defaults to
108 *32 filters.
109 */
110 static sptr
111 make(float rate, const std::vector<float>& taps, unsigned int filter_size = 32);
112
113 /*!
114 * Resets the filterbank's filter taps with the new prototype filter
115 * \param taps (vector/list of floats) The prototype filter to populate the
116 * filterbank.
117 */
118 virtual void set_taps(const std::vector<float>& taps) = 0;
119
120 /*!
121 * Return a vector<vector<>> of the filterbank taps
122 */
123 virtual std::vector<std::vector<float> > taps() const = 0;
124
125 /*!
126 * Print all of the filterbank taps to screen.
127 */
128 virtual void print_taps() = 0;
129
130 /*!
131 * Sets the resampling rate of the block.
132 */
133 virtual void set_rate(float rate) = 0;
134
135 /*!
136 * Sets the current phase offset in radians (0 to 2pi).
137 */
138 virtual void set_phase(float ph) = 0;
139
140 /*!
141 * Gets the current phase of the resampler in radians (2 to 2pi).
142 */
143 virtual float phase() const = 0;
144
145 /*!
146 * Gets the number of taps per filter.
147 */
148 virtual unsigned int taps_per_filter() const = 0;
149
150 /*!
151 * Gets the interpolation rate of the filter.
152 */
153 virtual unsigned int interpolation_rate() const = 0;
154
155 /*!
156 * Gets the decimation rate of the filter.
157 */
158 virtual unsigned int decimation_rate() const = 0;
159
160 /*!
161 * Gets the fractional rate of the filter.
162 */
163 virtual float fractional_rate() const = 0;
164
165 /*!
166 * Get the group delay of the filter.
167 */
168 virtual int group_delay() const = 0;
169
170 /*!
171 * Calculates the phase offset expected by a sine wave of
172 * frequency \p freq and sampling rate \p fs (assuming input
173 * sine wave has 0 degree phase).
174 */
175 virtual float phase_offset(float freq, float fs) = 0;
176};
177
178} /* namespace filter */
179} /* namespace gr */
180
181#endif /* INCLUDED_PFB_ARB_RESAMPLER_FFF_H */
The abstract base class for all 'terminal' processing blocks.
Definition: block.h:66
Polyphase filterbank arbitrary resampler with float input, float output and float taps.
Definition: pfb_arb_resampler_fff.h:96
virtual float phase() const =0
virtual unsigned int interpolation_rate() const =0
virtual float phase_offset(float freq, float fs)=0
virtual void set_rate(float rate)=0
virtual void set_taps(const std::vector< float > &taps)=0
virtual float fractional_rate() const =0
boost::shared_ptr< pfb_arb_resampler_fff > sptr
Definition: pfb_arb_resampler_fff.h:99
virtual unsigned int taps_per_filter() const =0
virtual unsigned int decimation_rate() const =0
virtual void set_phase(float ph)=0
virtual std::vector< std::vector< float > > taps() const =0
virtual int group_delay() const =0
static sptr make(float rate, const std::vector< float > &taps, unsigned int filter_size=32)
#define FILTER_API
Definition: gr-filter/include/gnuradio/filter/api.h:30
static const float taps[NSTEPS+1][NTAPS]
Definition: interpolator_taps.h:9
Include this header to use the message passing features.
Definition: basic_block.h:45