Merge branch 'next' of http://gnuradio.org/git/gnuradio into digital

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: