diff options
Diffstat (limited to 'gnuradio-core/src')
11 files changed, 579 insertions, 57 deletions
diff --git a/gnuradio-core/src/lib/filter/Makefile.am b/gnuradio-core/src/lib/filter/Makefile.am index 48ec55a62..c314431bf 100644 --- a/gnuradio-core/src/lib/filter/Makefile.am +++ b/gnuradio-core/src/lib/filter/Makefile.am @@ -215,6 +215,7 @@ 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 \ @@ -307,6 +308,7 @@ 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 \ @@ -374,6 +376,7 @@ 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 2af7fcc5c..8c3bb9eb6 100644 --- a/gnuradio-core/src/lib/filter/filter.i +++ b/gnuradio-core/src/lib/filter/filter.i @@ -36,6 +36,7 @@ #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> @@ -57,6 +58,7 @@ %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); +}; diff --git a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.cc b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.cc index a939609f3..633c5be07 100644 --- a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.cc +++ b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.cc @@ -187,6 +187,12 @@ gr_pfb_clock_sync_ccf::get_beta() const return d_beta; } +float +gr_pfb_clock_sync_ccf::get_clock_rate() const +{ + return d_rate_f; +} + /******************************************************************* *******************************************************************/ diff --git a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.h b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.h index 0fd8ba35b..54ae889d7 100644 --- a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.h +++ b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.h @@ -315,6 +315,11 @@ public: */ float get_beta() const; + /*! + * \brief Returns the current clock rate + */ + float get_clock_rate() const; + /******************************************************************* *******************************************************************/ diff --git a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.i b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.i index 78b9a6589..92ad1661a 100644 --- a/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.i +++ b/gnuradio-core/src/lib/filter/gr_pfb_clock_sync_ccf.i @@ -63,5 +63,5 @@ class gr_pfb_clock_sync_ccf : public gr_block float get_damping_factor() const; float get_alpha() const; float get_beta() const; - + float get_clock_rate() const; }; diff --git a/gnuradio-core/src/python/gnuradio/blks2impl/fm_demod.py b/gnuradio-core/src/python/gnuradio/blks2impl/fm_demod.py index 1910b5011..55870513a 100644 --- a/gnuradio-core/src/python/gnuradio/blks2impl/fm_demod.py +++ b/gnuradio-core/src/python/gnuradio/blks2impl/fm_demod.py @@ -88,7 +88,7 @@ class demod_20k0f3e_cf(fm_demod_cf): fm_demod_cf.__init__(self, channel_rate, audio_decim, 5000, # Deviation 3000, # Audio passband frequency - 4000) # Audio stopband frequency + 4500) # Audio stopband frequency class demod_200kf3e_cf(fm_demod_cf): """ diff --git a/gnuradio-core/src/python/gnuradio/blks2impl/pfb_arb_resampler.py b/gnuradio-core/src/python/gnuradio/blks2impl/pfb_arb_resampler.py index 62f40582e..3aadf700b 100644 --- a/gnuradio-core/src/python/gnuradio/blks2impl/pfb_arb_resampler.py +++ b/gnuradio-core/src/python/gnuradio/blks2impl/pfb_arb_resampler.py @@ -73,3 +73,56 @@ class pfb_arb_resampler_ccf(gr.hier_block2): def set_rate(self, rate): self.pfb.set_rate(rate) + + +class pfb_arb_resampler_fff(gr.hier_block2): + ''' + Convenience wrapper for the polyphase filterbank arbitrary resampler. + + The block takes a single float stream in and outputs a single float + stream out. As such, it requires no extra glue to handle the input/output + streams. This block is provided to be consistent with the interface to the + other PFB block. + ''' + def __init__(self, rate, taps=None, flt_size=32, atten=100): + gr.hier_block2.__init__(self, "pfb_arb_resampler_fff", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + self._rate = rate + self._size = flt_size + + if taps is not None: + self._taps = taps + else: + # Create a filter that covers the full bandwidth of the input signal + bw = 0.4 + tb = 0.2 + ripple = 0.1 + #self._taps = gr.firdes.low_pass_2(self._size, self._size, bw, tb, atten) + made = False + while not made: + try: + self._taps = optfir.low_pass(self._size, self._size, bw, bw+tb, ripple, atten) + made = True + except RuntimeError: + ripple += 0.01 + made = False + print("Warning: set ripple to %.4f dB. If this is a problem, adjust the attenuation or create your own filter taps." % (ripple)) + + # Build in an exit strategy; if we've come this far, it ain't working. + if(ripple >= 1.0): + raise RuntimeError("optfir could not generate an appropriate filter.") + + self.pfb = gr.pfb_arb_resampler_fff(self._rate, self._taps, self._size) + #print "PFB has %d taps\n" % (len(self._taps),) + + self.connect(self, self.pfb) + self.connect(self.pfb, self) + + # Note -- set_taps not implemented in base class yet + def set_taps(self, taps): + self.pfb.set_taps(taps) + + def set_rate(self, rate): + self.pfb.set_rate(rate) diff --git a/gnuradio-core/src/python/gnuradio/blks2impl/wfm_rcv_fmdet.py b/gnuradio-core/src/python/gnuradio/blks2impl/wfm_rcv_fmdet.py index 858b9cde6..3a93a11d6 100755 --- a/gnuradio-core/src/python/gnuradio/blks2impl/wfm_rcv_fmdet.py +++ b/gnuradio-core/src/python/gnuradio/blks2impl/wfm_rcv_fmdet.py @@ -28,8 +28,9 @@ class wfm_rcv_fmdet(gr.hier_block2): """ Hierarchical block for demodulating a broadcast FM signal. - The input is the downconverted complex baseband signal (gr_complex). - The output is two streams of the demodulated audio (float) 0=Left, 1=Right. + The input is the downconverted complex baseband signal + (gr_complex). The output is two streams of the demodulated + audio (float) 0=Left, 1=Right. @param demod_rate: input sample rate of complex baseband input. @type demod_rate: float @@ -39,16 +40,15 @@ class wfm_rcv_fmdet(gr.hier_block2): gr.hier_block2.__init__(self, "wfm_rcv_fmdet", gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature gr.io_signature(2, 2, gr.sizeof_float)) # Output signature - lowfreq = -125e3 - highfreq = 125e3 + lowfreq = -125e3/demod_rate + highfreq = 125e3/demod_rate audio_rate = demod_rate / audio_decimation - - # We assign to self so that outsiders can grab the demodulator + # We assign to self so that outsiders can grab the demodulator # if they need to. E.g., to plot its output. # # input: complex; output: float - + self.fm_demod = gr.fmdet_cf (demod_rate, lowfreq, highfreq, 0.05) # input: float; output: float @@ -62,25 +62,31 @@ class wfm_rcv_fmdet(gr.hier_block2): 15000 , width_of_transition_band, gr.firdes.WIN_HAMMING) + # input: float; output: float self.audio_filter = gr.fir_filter_fff (audio_decimation, audio_coeffs) if 1: - # Pick off the stereo carrier/2 with this filter. It attenuated 10 dB so apply 10 dB gain - # We pick off the negative frequency half because we want to base band by it! - ## NOTE THIS WAS HACKED TO OFFSET INSERTION LOSS DUE TO DEEMPHASIS + # Pick off the stereo carrier/2 with this filter. It + # attenuated 10 dB so apply 10 dB gain We pick off the + # negative frequency half because we want to base band by + # it! + ## NOTE THIS WAS HACKED TO OFFSET INSERTION LOSS DUE TO + ## DEEMPHASIS stereo_carrier_filter_coeffs = gr.firdes.complex_band_pass(10.0, - demod_rate, - -19020, - -18980, - width_of_transition_band, - gr.firdes.WIN_HAMMING) + demod_rate, + -19020, + -18980, + width_of_transition_band, + gr.firdes.WIN_HAMMING) #print "len stereo carrier filter = ",len(stereo_carrier_filter_coeffs) #print "stereo carrier filter ", stereo_carrier_filter_coeffs #print "width of transition band = ",width_of_transition_band, " audio rate = ", audio_rate - # Pick off the double side band suppressed carrier Left-Right audio. It is attenuated 10 dB so apply 10 dB gain + # Pick off the double side band suppressed carrier + # Left-Right audio. It is attenuated 10 dB so apply 10 dB + # gain stereo_dsbsc_filter_coeffs = gr.firdes.complex_band_pass(20.0, demod_rate, @@ -90,101 +96,121 @@ class wfm_rcv_fmdet(gr.hier_block2): gr.firdes.WIN_HAMMING) #print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs) #print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs - # construct overlap add filter system from coefficients for stereo carrier - self.stereo_carrier_filter = gr.fir_filter_fcc(audio_decimation, stereo_carrier_filter_coeffs) - - # carrier is twice the picked off carrier so arrange to do a commplex multiply + # construct overlap add filter system from coefficients + # for stereo carrier + self.stereo_carrier_filter = gr.fir_filter_fcc(audio_decimation, + stereo_carrier_filter_coeffs) + # carrier is twice the picked off carrier so arrange to do + # a commplex multiply self.stereo_carrier_generator = gr.multiply_cc(); # Pick off the rds signal - stereo_rds_filter_coeffs = gr.firdes.complex_band_pass(30.0, - demod_rate, - 57000 - 1500, - 57000 + 1500, - width_of_transition_band, - gr.firdes.WIN_HAMMING) + demod_rate, + 57000 - 1500, + 57000 + 1500, + width_of_transition_band, + gr.firdes.WIN_HAMMING) #print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs) #print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs # construct overlap add filter system from coefficients for stereo carrier - self.rds_signal_filter = gr.fir_filter_fcc(audio_decimation, stereo_rds_filter_coeffs) - - - - - - + self.rds_signal_filter = gr.fir_filter_fcc(audio_decimation, + stereo_rds_filter_coeffs) self.rds_carrier_generator = gr.multiply_cc(); self.rds_signal_generator = gr.multiply_cc(); self_rds_signal_processor = gr.null_sink(gr.sizeof_gr_complex); - - alpha = 5 * 0.25 * math.pi / (audio_rate) beta = alpha * alpha / 4.0 max_freq = -2.0*math.pi*18990/audio_rate; - min_freq = -2.0*math.pi*19010/audio_rate; + min_freq = -2.0*math.pi*19010/audio_rate; + self.stereo_carrier_pll_recovery = gr.pll_refout_cc(alpha,beta, + max_freq, + min_freq); + + #self.stereo_carrier_pll_recovery.squelch_enable(False) + ##pll_refout does not have squelch yet, so disabled for + #now - self.stereo_carrier_pll_recovery = gr.pll_refout_cc(alpha,beta,max_freq,min_freq); - #self.stereo_carrier_pll_recovery.squelch_enable(False) #pll_refout does not have squelch yet, so disabled for now - - - # set up mixer (multiplier) to get the L-R signal at baseband + # set up mixer (multiplier) to get the L-R signal at + # baseband self.stereo_basebander = gr.multiply_cc(); - # pick off the real component of the basebanded L-R signal. The imaginary SHOULD be zero + # pick off the real component of the basebanded L-R + # signal. The imaginary SHOULD be zero self.LmR_real = gr.complex_to_real(); self.Make_Left = gr.add_ff(); self.Make_Right = gr.sub_ff(); - self.stereo_dsbsc_filter = gr.fir_filter_fcc(audio_decimation, stereo_dsbsc_filter_coeffs) + self.stereo_dsbsc_filter = gr.fir_filter_fcc(audio_decimation, + stereo_dsbsc_filter_coeffs) if 1: - # send the real signal to complex filter to pick off the carrier and then to one side of a multiplier - self.connect (self, self.fm_demod,self.stereo_carrier_filter,self.stereo_carrier_pll_recovery, (self.stereo_carrier_generator,0)) + # send the real signal to complex filter to pick off the + # carrier and then to one side of a multiplier + self.connect (self, self.fm_demod, self.stereo_carrier_filter, + self.stereo_carrier_pll_recovery, + (self.stereo_carrier_generator,0)) + # send the already filtered carrier to the otherside of the carrier + # the resulting signal from this multiplier is the carrier + # with correct phase but at -38000 Hz. self.connect (self.stereo_carrier_pll_recovery, (self.stereo_carrier_generator,1)) - # the resulting signal from this multiplier is the carrier with correct phase but at -38000 Hz. # send the new carrier to one side of the mixer (multiplier) self.connect (self.stereo_carrier_generator, (self.stereo_basebander,0)) + # send the demphasized audio to the DSBSC pick off filter, the complex # DSBSC signal at +38000 Hz is sent to the other side of the mixer/multiplier - self.connect (self.fm_demod,self.stereo_dsbsc_filter, (self.stereo_basebander,1)) # the result is BASEBANDED DSBSC with phase zero! + self.connect (self.fm_demod,self.stereo_dsbsc_filter, (self.stereo_basebander,1)) - # Pick off the real part since the imaginary is theoretically zero and then to one side of a summer + # Pick off the real part since the imaginary is + # theoretically zero and then to one side of a summer self.connect (self.stereo_basebander, self.LmR_real, (self.Make_Left,0)) - #take the same real part of the DSBSC baseband signal and send it to negative side of a subtracter + + #take the same real part of the DSBSC baseband signal and + #send it to negative side of a subtracter self.connect (self.LmR_real,(self.Make_Right,1)) - # Make rds carrier by taking the squared pilot tone and multiplying by pilot tone + # Make rds carrier by taking the squared pilot tone and + # multiplying by pilot tone self.connect (self.stereo_basebander,(self.rds_carrier_generator,0)) self.connect (self.stereo_carrier_pll_recovery,(self.rds_carrier_generator,1)) - # take signal, filter off rds, send into mixer 0 channel + + # take signal, filter off rds, send into mixer 0 channel self.connect (self.fm_demod,self.rds_signal_filter,(self.rds_signal_generator,0)) - # take rds_carrier_generator output and send into mixer 1 channel + + # take rds_carrier_generator output and send into mixer 1 + # channel self.connect (self.rds_carrier_generator,(self.rds_signal_generator,1)) - # send basebanded rds signal and send into "processor" which for now is a null sink + + # send basebanded rds signal and send into "processor" + # which for now is a null sink self.connect (self.rds_signal_generator,self_rds_signal_processor) if 1: - # pick off the audio, L+R that is what we used to have and send it to the summer + # pick off the audio, L+R that is what we used to have and + # send it to the summer self.connect(self.fm_demod, self.audio_filter, (self.Make_Left, 1)) - # take the picked off L+R audio and send it to the PLUS side of the subtractor + + # take the picked off L+R audio and send it to the PLUS + # side of the subtractor self.connect(self.audio_filter,(self.Make_Right, 0)) + # The result of Make_Left gets (L+R) + (L-R) and results in 2*L # The result of Make_Right gets (L+R) - (L-R) and results in 2*R self.connect(self.Make_Left , self.deemph_Left, (self, 0)) self.connect(self.Make_Right, self.deemph_Right, (self, 1)) + # NOTE: mono support will require variable number of outputs in hier_block2s # See ticket:174 in Trac database #else: |