diff options
| -rw-r--r-- | gnuradio-core/src/lib/filter/Makefile.am | 3 | ||||
| -rw-r--r-- | gnuradio-core/src/lib/filter/filter.i | 2 | ||||
| -rw-r--r-- | gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.cc | 209 | ||||
| -rw-r--r-- | gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.h | 176 | ||||
| -rw-r--r-- | gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.i | 42 |
5 files changed, 432 insertions, 0 deletions
diff --git a/gnuradio-core/src/lib/filter/Makefile.am b/gnuradio-core/src/lib/filter/Makefile.am index d8f634c38..b1950ce2b 100644 --- a/gnuradio-core/src/lib/filter/Makefile.am +++ b/gnuradio-core/src/lib/filter/Makefile.am @@ -216,6 +216,7 @@ libfilter_la_common_SOURCES = \ gr_pfb_decimator_ccf.cc \ gr_pfb_interpolator_ccf.cc \ gr_pfb_arb_resampler_ccf.cc \ + gr_pfb_arb_resampler_fff.cc \ gr_pfb_clock_sync_ccf.cc \ gr_pfb_clock_sync_fff.cc \ gr_dc_blocker_cc.cc \ @@ -309,6 +310,7 @@ grinclude_HEADERS = \ gr_pfb_decimator_ccf.h \ gr_pfb_interpolator_ccf.h \ gr_pfb_arb_resampler_ccf.h \ + gr_pfb_arb_resampler_fff.h \ gr_pfb_clock_sync_ccf.h \ gr_pfb_clock_sync_fff.h \ gr_dc_blocker_cc.h \ @@ -377,6 +379,7 @@ swiginclude_HEADERS = \ gr_pfb_decimator_ccf.i \ gr_pfb_interpolator_ccf.i \ gr_pfb_arb_resampler_ccf.i \ + gr_pfb_arb_resampler_fff.i \ gr_pfb_clock_sync_ccf.i \ gr_pfb_clock_sync_fff.i \ gr_dc_blocker_cc.i \ diff --git a/gnuradio-core/src/lib/filter/filter.i b/gnuradio-core/src/lib/filter/filter.i index 58bb4f0d5..a93a9f6dd 100644 --- a/gnuradio-core/src/lib/filter/filter.i +++ b/gnuradio-core/src/lib/filter/filter.i @@ -37,6 +37,7 @@ #include <gr_pfb_decimator_ccf.h> #include <gr_pfb_interpolator_ccf.h> #include <gr_pfb_arb_resampler_ccf.h> +#include <gr_pfb_arb_resampler_fff.h> #include <gr_pfb_clock_sync_ccf.h> #include <gr_pfb_clock_sync_fff.h> #include <gr_dc_blocker_cc.h> @@ -59,6 +60,7 @@ %include "gr_pfb_decimator_ccf.i" %include "gr_pfb_interpolator_ccf.i" %include "gr_pfb_arb_resampler_ccf.i" +%include "gr_pfb_arb_resampler_fff.i" %include "gr_pfb_decimator_ccf.i" %include "gr_pfb_interpolator_ccf.i" %include "gr_pfb_arb_resampler_ccf.i" diff --git a/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.cc b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.cc new file mode 100644 index 000000000..9035e67f4 --- /dev/null +++ b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.cc @@ -0,0 +1,209 @@ +/* -*- c++ -*- */ +/* + * Copyright 2009-2011 Free Software Foundation, Inc. + * + * This file is part of GNU Radio + * + * GNU Radio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 3, or (at your option) + * any later version. + * + * GNU Radio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with GNU Radio; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 51 Franklin Street, + * Boston, MA 02110-1301, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gr_pfb_arb_resampler_fff.h> +#include <gr_fir_fff.h> +#include <gr_fir_util.h> +#include <gr_io_signature.h> +#include <cstdio> + +gr_pfb_arb_resampler_fff_sptr gr_make_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size) +{ + return gnuradio::get_initial_sptr(new gr_pfb_arb_resampler_fff (rate, taps, + filter_size)); +} + + +gr_pfb_arb_resampler_fff::gr_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size) + : gr_block ("pfb_arb_resampler_fff", + gr_make_io_signature (1, 1, sizeof(float)), + gr_make_io_signature (1, 1, sizeof(float))), + d_updated (false) +{ + d_acc = 0; // start accumulator at 0 + + /* The number of filters is specified by the user as the filter size; + this is also the interpolation rate of the filter. We use it and the + rate provided to determine the decimation rate. This acts as a + rational resampler. The flt_rate is calculated as the residual + between the integer decimation rate and the real decimation rate and + will be used to determine to interpolation point of the resampling + process. + */ + d_int_rate = filter_size; + set_rate(rate); + + // Store the last filter between calls to work + d_last_filter = 0; + + d_start_index = 0; + + d_filters = std::vector<gr_fir_fff*>(d_int_rate); + d_diff_filters = std::vector<gr_fir_fff*>(d_int_rate); + + // Create an FIR filter for each channel and zero out the taps + std::vector<float> vtaps(0, d_int_rate); + for(unsigned int i = 0; i < d_int_rate; i++) { + d_filters[i] = gr_fir_util::create_gr_fir_fff(vtaps); + d_diff_filters[i] = gr_fir_util::create_gr_fir_fff(vtaps); + } + + // Now, actually set the filters' taps + std::vector<float> dtaps; + create_diff_taps(taps, dtaps); + create_taps(taps, d_taps, d_filters); + create_taps(dtaps, d_dtaps, d_diff_filters); +} + +gr_pfb_arb_resampler_fff::~gr_pfb_arb_resampler_fff () +{ + for(unsigned int i = 0; i < d_int_rate; i++) { + delete d_filters[i]; + } +} + +void +gr_pfb_arb_resampler_fff::create_taps (const std::vector<float> &newtaps, + std::vector< std::vector<float> > &ourtaps, + std::vector<gr_fir_fff*> &ourfilter) +{ + unsigned int ntaps = newtaps.size(); + d_taps_per_filter = (unsigned int)ceil((double)ntaps/(double)d_int_rate); + + // Create d_numchan vectors to store each channel's taps + ourtaps.resize(d_int_rate); + + // Make a vector of the taps plus fill it out with 0's to fill + // each polyphase filter with exactly d_taps_per_filter + std::vector<float> tmp_taps; + tmp_taps = newtaps; + while((float)(tmp_taps.size()) < d_int_rate*d_taps_per_filter) { + tmp_taps.push_back(0.0); + } + + // Partition the filter + for(unsigned int i = 0; i < d_int_rate; i++) { + // Each channel uses all d_taps_per_filter with 0's if not enough taps to fill out + ourtaps[d_int_rate-1-i] = std::vector<float>(d_taps_per_filter, 0); + for(unsigned int j = 0; j < d_taps_per_filter; j++) { + ourtaps[d_int_rate - 1 - i][j] = tmp_taps[i + j*d_int_rate]; + } + + // Build a filter for each channel and add it's taps to it + ourfilter[i]->set_taps(ourtaps[d_int_rate-1-i]); + } + + // Set the history to ensure enough input items for each filter + set_history (d_taps_per_filter + 1); + + d_updated = true; +} + +void +gr_pfb_arb_resampler_fff::create_diff_taps(const std::vector<float> &newtaps, + std::vector<float> &difftaps) +{ + // Calculate the differential taps (derivative filter) by taking the difference + // between two taps. Duplicate the last one to make both filters the same length. + float tap; + difftaps.clear(); + for(unsigned int i = 0; i < newtaps.size()-1; i++) { + tap = newtaps[i+1] - newtaps[i]; + difftaps.push_back(tap); + } + difftaps.push_back(tap); +} + +void +gr_pfb_arb_resampler_fff::print_taps() +{ + unsigned int i, j; + for(i = 0; i < d_int_rate; i++) { + printf("filter[%d]: [", i); + for(j = 0; j < d_taps_per_filter; j++) { + printf(" %.4e", d_taps[i][j]); + } + printf("]\n"); + } +} + +int +gr_pfb_arb_resampler_fff::general_work (int noutput_items, + gr_vector_int &ninput_items, + gr_vector_const_void_star &input_items, + gr_vector_void_star &output_items) +{ + float *in = (float *) input_items[0]; + float *out = (float *) output_items[0]; + + if (d_updated) { + d_updated = false; + return 0; // history requirements may have changed. + } + + int i = 0, count = d_start_index; + unsigned int j; + float o0, o1; + + // Restore the last filter position + j = d_last_filter; + + // produce output as long as we can and there are enough input samples + int max_input = ninput_items[0]-(int)d_taps_per_filter; + while((i < noutput_items) && (count < max_input)) { + // start j by wrapping around mod the number of channels + while((j < d_int_rate) && (i < noutput_items)) { + // Take the current filter and derivative filter output + o0 = d_filters[j]->filter(&in[count]); + o1 = d_diff_filters[j]->filter(&in[count]); + + out[i] = o0 + o1*d_acc; // linearly interpolate between samples + i++; + + // Adjust accumulator and index into filterbank + d_acc += d_flt_rate; + j += d_dec_rate + (int)floor(d_acc); + d_acc = fmodf(d_acc, 1.0); + } + if(i < noutput_items) { // keep state for next entry + float ss = (int)(j / d_int_rate); // number of items to skip ahead by + count += ss; // we have fully consumed another input + j = j % d_int_rate; // roll filter around + } + } + + // Store the current filter position and start of next sample + d_last_filter = j; + d_start_index = std::max(0, count - ninput_items[0]); + + // consume all we've processed but no more than we can + consume_each(std::min(count, ninput_items[0])); + return i; +} diff --git a/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.h b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.h new file mode 100644 index 000000000..541df8aa4 --- /dev/null +++ b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.h @@ -0,0 +1,176 @@ +/* -*- c++ -*- */ +/* + * Copyright 2009-2011 Free Software Foundation, Inc. + * + * This file is part of GNU Radio + * + * GNU Radio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 3, or (at your option) + * any later version. + * + * GNU Radio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with GNU Radio; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 51 Franklin Street, + * Boston, MA 02110-1301, USA. + */ + + +#ifndef INCLUDED_GR_PFB_ARB_RESAMPLER_FFF_H +#define INCLUDED_GR_PFB_ARB_RESAMPLER_FFF_H + +#include <gr_block.h> + +class gr_pfb_arb_resampler_fff; +typedef boost::shared_ptr<gr_pfb_arb_resampler_fff> gr_pfb_arb_resampler_fff_sptr; +gr_pfb_arb_resampler_fff_sptr gr_make_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size=32); + +class gr_fir_fff; + +/*! + * \class gr_pfb_arb_resampler_fff + * + * \brief Polyphase filterbank arbitrary resampler with + * float input, float output and float taps + * + * \ingroup filter_blk + * + * This block takes in a signal stream and performs arbitrary + * resampling. The resampling rate can be any real + * number <EM>r</EM>. The resampling is done by constructing + * <EM>N</EM> filters where <EM>N</EM> is the interpolation rate. We + * then calculate <EM>D</EM> where <EM>D = floor(N/r)</EM>. + * + * Using <EM>N</EM> and <EM>D</EM>, we can perform rational resampling + * where <EM>N/D</EM> is a rational number close to the input rate + * <EM>r</EM> where we have <EM>N</EM> filters and we cycle through + * them as a polyphase filterbank with a stride of <EM>D</EM> so that + * <EM>i+1 = (i + D) % N</EM>. + * + * To get the arbitrary rate, we want to interpolate between two + * points. For each value out, we take an output from the current + * filter, <EM>i</EM>, and the next filter <EM>i+1</EM> 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, <EM>N</EM>, used determines the + * quantization error; the larger <EM>N</EM>, the smaller the + * noise. You can design for a specified noise floor by setting the + * filter size (parameters <EM>filter_size</EM>). 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 <EM>fs</EM>, a 3-dB bandwidth of + * <EM>BW</EM> and a transition bandwidth of <EM>TB</EM>. We can also + * specify the out-of-band attenuation to use, <EM>ATT</EM>, 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 (<EM>32</EM>). + * + * <B><EM>self._taps = gr.firdes.low_pass_2(32, 32*fs, BW, TB, + * attenuation_dB=ATT, window=gr.firdes.WIN_BLACKMAN_hARRIS)</EM></B> + * + * The theory behind this block can be found in Chapter 7.5 of + * the following book. + * + * <B><EM>f. harris, "Multirate Signal Processing for Communication + * Systems", Upper Saddle River, NJ: Prentice Hall, Inc. 2004.</EM></B> + */ + +class gr_pfb_arb_resampler_fff : public gr_block +{ + private: + /*! + * Build the polyphase filterbank arbitray resampler. + * \param rate (float) Specifies the resampling rate to use + * \param taps (vector/list of floats) The prototype filter to populate the filterbank. The taps + * should be generated at the filter_size sampling rate. + * \param 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. + */ + friend gr_pfb_arb_resampler_fff_sptr gr_make_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size); + + std::vector<gr_fir_fff*> d_filters; + std::vector<gr_fir_fff*> d_diff_filters; + std::vector< std::vector<float> > d_taps; + std::vector< std::vector<float> > d_dtaps; + unsigned int d_int_rate; // the number of filters (interpolation rate) + unsigned int d_dec_rate; // the stride through the filters (decimation rate) + float d_flt_rate; // residual rate for the linear interpolation + float d_acc; + unsigned int d_last_filter; + int d_start_index; + unsigned int d_taps_per_filter; + bool d_updated; + + /*! + * Build the polyphase filterbank arbitray resampler. + * \param rate (float) Specifies the resampling rate to use + * \param taps (vector/list of floats) The prototype filter to populate the filterbank. The taps + * should be generated at the filter_size sampling rate. + * \param 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. + */ + gr_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size); + + void create_diff_taps(const std::vector<float> &newtaps, + std::vector<float> &difftaps); + + /*! + * Resets the filterbank's filter taps with the new prototype filter + * \param newtaps (vector of floats) The prototype filter to populate the filterbank. + * The taps should be generated at the interpolated sampling rate. + * \param ourtaps (vector of floats) Reference to our internal member of holding the taps. + * \param ourfilter (vector of filters) Reference to our internal filter to set the taps for. + */ + void create_taps (const std::vector<float> &newtaps, + std::vector< std::vector<float> > &ourtaps, + std::vector<gr_fir_fff*> &ourfilter); + + +public: + ~gr_pfb_arb_resampler_fff (); + + // FIXME: See about a set_taps function during runtime. + + /*! + * Print all of the filterbank taps to screen. + */ + void print_taps(); + void set_rate (float rate) { + d_dec_rate = (unsigned int)floor(d_int_rate/rate); + d_flt_rate = (d_int_rate/rate) - d_dec_rate; + set_relative_rate(rate); + } + + int general_work (int noutput_items, + gr_vector_int &ninput_items, + gr_vector_const_void_star &input_items, + gr_vector_void_star &output_items); +}; + +#endif diff --git a/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.i b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.i new file mode 100644 index 000000000..8c1db22c3 --- /dev/null +++ b/gnuradio-core/src/lib/filter/gr_pfb_arb_resampler_fff.i @@ -0,0 +1,42 @@ +/* -*- c++ -*- */ +/* + * Copyright 2009,2011 Free Software Foundation, Inc. + * + * This file is part of GNU Radio + * + * GNU Radio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 3, or (at your option) + * any later version. + * + * GNU Radio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with GNU Radio; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 51 Franklin Street, + * Boston, MA 02110-1301, USA. + */ + +GR_SWIG_BLOCK_MAGIC(gr,pfb_arb_resampler_fff); + +gr_pfb_arb_resampler_fff_sptr gr_make_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size=32); + +class gr_pfb_arb_resampler_fff : public gr_block +{ + private: + gr_pfb_arb_resampler_fff (float rate, + const std::vector<float> &taps, + unsigned int filter_size); + + public: + ~gr_pfb_arb_resampler_fff (); + + //void set_taps (const std::vector<float> &taps); + void print_taps(); + void set_rate (float rate); +}; |