/* -*- c++ -*- */
/*
- * Copyright 2009 Free Software Foundation, Inc.
+ * Copyright 2009,2010 Free Software Foundation, Inc.
*
* This file is part of GNU Radio
*
#include <cstring>
gr_pfb_channelizer_ccf_sptr gr_make_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps)
+ const std::vector<float> &taps,
+ float oversample_rate)
{
- return gr_pfb_channelizer_ccf_sptr (new gr_pfb_channelizer_ccf (numchans, taps));
+ return gr_pfb_channelizer_ccf_sptr (new gr_pfb_channelizer_ccf (numchans, taps,
+ oversample_rate));
}
gr_pfb_channelizer_ccf::gr_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps)
- : gr_sync_block ("pfb_channelizer_ccf",
- gr_make_io_signature (numchans, numchans, sizeof(gr_complex)),
- gr_make_io_signature (1, 1, numchans*sizeof(gr_complex))),
- d_updated (false)
+ const std::vector<float> &taps,
+ float oversample_rate)
+ : gr_block ("pfb_channelizer_ccf",
+ gr_make_io_signature (numchans, numchans, sizeof(gr_complex)),
+ gr_make_io_signature (1, 1, numchans*sizeof(gr_complex))),
+ d_updated (false), d_numchans(numchans), d_oversample_rate(oversample_rate)
{
- d_numchans = numchans;
+ // The over sampling rate must be rationally related to the number of channels
+ // in that it must be N/i for i in [1,N], which gives an outputsample rate
+ // of [fs/N, fs] where fs is the input sample rate.
+ // This tests the specified input sample rate to see if it conforms to this
+ // requirement within a few significant figures.
+ double intp = 0;
+ double x = (10000.0*rint(numchans / oversample_rate)) / 10000.0;
+ double fltp = modf(numchans / oversample_rate, &intp);
+ if(fltp != 0.0)
+ throw std::invalid_argument("gr_pfb_channelizer: oversample rate must be N/i for i in [1, N]");
+
d_filters = std::vector<gr_fir_ccf*>(d_numchans);
// Create an FIR filter for each channel and zero out the taps
// Create the FFT to handle the output de-spinning of the channels
d_fft = new gri_fft_complex (d_numchans, false);
+
+ // Although the filters change, we use this look up table
+ // to set the index of the FFT input buffer, which equivalently
+ // performs the FFT shift operation on every other turn.
+ d_rate_ratio = (int)rintf(d_numchans / d_oversample_rate);
+ d_idxlut = new int[d_numchans];
+ for(unsigned int i = 0; i < d_numchans; i++) {
+ d_idxlut[i] = d_numchans - ((i + d_rate_ratio) % d_numchans) - 1;
+ }
+
+ // Calculate the number of filtering rounds to do to evenly
+ // align the input vectors with the output channels
+ d_output_multiple = 1;
+ while((d_output_multiple * d_rate_ratio) % d_numchans != 0)
+ d_output_multiple++;
+ set_output_multiple(d_output_multiple);
}
gr_pfb_channelizer_ccf::~gr_pfb_channelizer_ccf ()
{
+ delete [] d_idxlut;
+
for(unsigned int i = 0; i < d_numchans; i++) {
delete d_filters[i];
}
}
// Set the history to ensure enough input items for each filter
- set_history (d_taps_per_filter);
+ set_history (d_taps_per_filter+1);
d_updated = true;
}
int
-gr_pfb_channelizer_ccf::work (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
+gr_pfb_channelizer_ccf::general_work (int noutput_items,
+ gr_vector_int &ninput_items,
+ gr_vector_const_void_star &input_items,
+ gr_vector_void_star &output_items)
{
gr_complex *in = (gr_complex *) input_items[0];
gr_complex *out = (gr_complex *) output_items[0];
return 0; // history requirements may have changed.
}
- for(int i = 0; i < noutput_items; i++) {
- // Move through filters from bottom to top
- for(int j = d_numchans-1; j >= 0; j--) {
- // Take in the items from the first input stream to d_numchans
- in = (gr_complex*)input_items[d_numchans - 1 - j];
+ int n=1, i=-1, j=0, last;
+ int toconsume = (int)rintf(noutput_items/d_oversample_rate);
+ while(n <= toconsume) {
+ j = 0;
+ i = (i + d_rate_ratio) % d_numchans;
+ last = i;
+ while(i >= 0) {
+ in = (gr_complex*)input_items[j];
+ d_fft->get_inbuf()[d_idxlut[j]] = d_filters[i]->filter(&in[n]);
+ j++;
+ i--;
+ }
- // Filter current input stream from bottom filter to top
- d_fft->get_inbuf()[j] = d_filters[j]->filter(&in[i]);
+ i = d_numchans-1;
+ while(i > last) {
+ in = (gr_complex*)input_items[j];
+ d_fft->get_inbuf()[d_idxlut[j]] = d_filters[i]->filter(&in[n-1]);
+ j++;
+ i--;
}
+ n += (i+d_rate_ratio) >= (int)d_numchans;
+
// despin through FFT
d_fft->execute();
- memcpy(&out[d_numchans*i], d_fft->get_outbuf(), d_numchans*sizeof(gr_complex));
+ memcpy(out, d_fft->get_outbuf(), d_numchans*sizeof(gr_complex));
+ out += d_numchans;
}
-
+
+ consume_each(toconsume);
return noutput_items;
}
/* -*- c++ -*- */
/*
- * Copyright 2009 Free Software Foundation, Inc.
+ * Copyright 2009,2010 Free Software Foundation, Inc.
*
* This file is part of GNU Radio
*
#ifndef INCLUDED_GR_PFB_CHANNELIZER_CCF_H
#define INCLUDED_GR_PFB_CHANNELIZER_CCF_H
-#include <gr_sync_block.h>
+#include <gr_block.h>
class gr_pfb_channelizer_ccf;
typedef boost::shared_ptr<gr_pfb_channelizer_ccf> gr_pfb_channelizer_ccf_sptr;
gr_pfb_channelizer_ccf_sptr gr_make_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps);
+ const std::vector<float> &taps,
+ float oversample_rate=1);
class gr_fir_ccf;
class gri_fft_complex;
* <B><EM>self._taps = gr.firdes.low_pass_2(1, fs, BW, TB,
* attenuation_dB=ATT, window=gr.firdes.WIN_BLACKMAN_hARRIS)</EM></B>
*
+ * The filter output can also be overs ampled. The over sampling rate
+ * is the ratio of the the actual output sampling rate to the normal
+ * output sampling rate. It must be rationally related to the number
+ * of channels as N/i for i in [1,N], which gives an outputsample rate
+ * of [fs/N, fs] where fs is the input sample rate and N is the number
+ * of channels.
+ *
+ * For example, for 6 channels with fs = 6000 Hz, the normal rate is
+ * 6000/6 = 1000 Hz. Allowable oversampling rates are 6/6, 6/5, 6/4,
+ * 6/3, 6/2, and 6/1 where the output sample rate of a 6/1 oversample
+ * ratio is 6000 Hz, or 6 times the normal 1000 Hz. A rate of 6/5 = 1.2,
+ * so the output rate would be 1200 Hz.
+ *
* The theory behind this block can be found in Chapter 6 of
* the following book.
*
*
*/
-class gr_pfb_channelizer_ccf : public gr_sync_block
+class gr_pfb_channelizer_ccf : public gr_block
{
private:
/*!
* Build the polyphase filterbank decimator.
* \param numchans (unsigned integer) Specifies the number of channels <EM>M</EM>
* \param taps (vector/list of floats) The prototype filter to populate the filterbank.
+ * \param oversample_rate (float) The over sampling rate is the ratio of the the actual
+ * output sampling rate to the normal output sampling rate.
+ * It must be rationally related to the number of channels
+ * as N/i for i in [1,N], which gives an outputsample rate
+ * of [fs/N, fs] where fs is the input sample rate and N is
+ * the number of channels.
+ *
+ * For example, for 6 channels with fs = 6000 Hz, the normal
+ * rate is 6000/6 = 1000 Hz. Allowable oversampling rates
+ * are 6/6, 6/5, 6/4, 6/3, 6/2, and 6/1 where the output
+ * sample rate of a 6/1 oversample ratio is 6000 Hz, or
+ * 6 times the normal 1000 Hz.
*/
friend gr_pfb_channelizer_ccf_sptr gr_make_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps);
+ const std::vector<float> &taps,
+ float oversample_rate);
+ bool d_updated;
+ unsigned int d_numchans;
+ float d_oversample_rate;
std::vector<gr_fir_ccf*> d_filters;
std::vector< std::vector<float> > d_taps;
- gri_fft_complex *d_fft;
- unsigned int d_numchans;
unsigned int d_taps_per_filter;
- bool d_updated;
+ gri_fft_complex *d_fft;
+ int *d_idxlut;
+ int d_rate_ratio;
+ int d_output_multiple;
/*!
* Build the polyphase filterbank decimator.
* \param taps (vector/list of floats) The prototype filter to populate the filterbank.
*/
gr_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps);
+ const std::vector<float> &taps,
+ float oversample_rate);
public:
~gr_pfb_channelizer_ccf ();
*/
void print_taps();
- int work (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items);
+ 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
/* -*- c++ -*- */
/*
- * Copyright 2009 Free Software Foundation, Inc.
+ * Copyright 2009,2010 Free Software Foundation, Inc.
*
* This file is part of GNU Radio
*
GR_SWIG_BLOCK_MAGIC(gr,pfb_channelizer_ccf);
gr_pfb_channelizer_ccf_sptr gr_make_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps);
+ const std::vector<float> &taps,
+ float oversample_rate=1);
-class gr_pfb_channelizer_ccf : public gr_sync_block
+class gr_pfb_channelizer_ccf : public gr_block
{
private:
gr_pfb_channelizer_ccf (unsigned int numchans,
- const std::vector<float> &taps);
+ const std::vector<float> &taps,
+ float oversample_rate);
public:
~gr_pfb_channelizer_ccf ();
#!/usr/bin/env python
#
-# Copyright 2009 Free Software Foundation, Inc.
+# Copyright 2009,2010 Free Software Foundation, Inc.
#
# This file is part of GNU Radio
#
This simplifies the interface by allowing a single input stream to connect to this block.
It will then output a stream for each channel.
'''
- def __init__(self, numchans, taps):
+ def __init__(self, numchans, taps, oversample_rate=1):
gr.hier_block2.__init__(self, "pfb_channelizer_ccf",
gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature
gr.io_signature(numchans, numchans, gr.sizeof_gr_complex)) # Output signature
self._numchans = numchans
self._taps = taps
+ self._oversample_rate = oversample_rate
self.s2ss = gr.stream_to_streams(gr.sizeof_gr_complex, self._numchans)
- self.pfb = gr.pfb_channelizer_ccf(self._numchans, self._taps)
+ self.pfb = gr.pfb_channelizer_ccf(self._numchans, self._taps,
+ self._oversample_rate)
self.v2s = gr.vector_to_streams(gr.sizeof_gr_complex, self._numchans)
self.connect(self, self.s2ss)
X,freq = mlab.psd(d, NFFT=fftlen, noverlap=fftlen/4, Fs=fs,
window = lambda d: d*winfunc(fftlen),
scale_by_freq=True)
- X_in = 10.0*scipy.log10(abs(fftpack.fftshift(X)))
+ X_in = 10.0*scipy.log10(abs(X))
f_in = scipy.arange(-fs/2.0, fs/2.0, fs/float(X_in.size))
pin_f = spin_f.plot(f_in, X_in, "b")
spin_f.set_xlim([min(f_in), max(f_in)+1])
X,freq = mlab.psd(d, NFFT=fftlen, noverlap=fftlen/4, Fs=fs_o,
window = lambda d: d*winfunc(fftlen),
scale_by_freq=True)
- X_o = 10.0*scipy.log10(abs(fftpack.fftshift(X)))
+ X_o = 10.0*scipy.log10(abs(X))
f_o = scipy.arange(-fs_o/2.0, fs_o/2.0, fs_o/float(X_o.size))
p2_f = sp1_f.plot(f_o, X_o, "b")
sp1_f.set_xlim([min(f_o), max(f_o)+1])
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