pfb_channelizer_ccf.h

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/* -*- c++ -*- */
/*
 * Copyright 2009,2010,2012 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_FILTER_PFB_CHANNELIZER_CCF_H
#define INCLUDED_FILTER_PFB_CHANNELIZER_CCF_H

#include <gnuradio/filter/api.h>
#include <gnuradio/block.h>

namespace gr {
  namespace filter {

    /*!
     * \brief Polyphase filterbank channelizer with
     *        gr_complex input, gr_complex output and float taps
     * \ingroup channelizers_blk
     *
     * \details
     * This block takes in complex inputs and channelizes it to <EM>M</EM>
     * channels of equal bandwidth. Each of the resulting channels is
     * decimated to the new rate that is the input sampling rate
     * <EM>fs</EM> divided by the number of channels, <EM>M</EM>.
     *
     * The PFB channelizer code takes the taps generated above and builds
     * a set of filters. The set contains <EM>M</EM> number of filters
     * and each filter contains ceil(taps.size()/decim) number of taps.
     * Each tap from the filter prototype is sequentially inserted into
     * the next filter. When all of the input taps are used, the remaining
     * filters in the filterbank are filled out with 0's to make sure each
     * filter has the same number of taps.
     *
     * Each filter operates using the gr::blocks::fir_filter_XXX
     * classs of GNU Radio, which takes the input stream at <EM>i</EM>
     * and performs the inner product calculation to <EM>i+(n-1)</EM>
     * where <EM>n</EM> is the number of filter taps. To efficiently
     * handle this in the GNU Radio structure, each filter input must
     * come from its own input stream. So the channelizer must be
     * provided with <EM>M</EM> streams where the input stream has
     * been deinterleaved. This is most easily done using the
     * gr::blocks::stream_to_streams block.
     *
     * The output is then produced as a vector, where index <EM>i</EM>
     * in the vector is the next sample from the <EM>i</EM>th
     * channel. This is most easily handled by sending the output to a
     * gr::blocks::vector_to_streams block to handle the conversion
     * and passing <EM>M</EM> streams out.
     *
     * The input and output formatting is done using a hier_block2 called
     * pfb_channelizer_ccf. This can take in a single stream and outputs
     * <EM>M</EM> streams based on the behavior described above.
     *
     * The filter's taps should be based on the input sampling rate.
     *
     * For example, using the GNU Radio's firdes utility to building
     * filters, 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 unity.
     *
     *      <B><EM>self._taps = filter.firdes.low_pass_2(1, fs, BW, TB,
     *           attenuation_dB=ATT, window=filter.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.
     *
     *    <B><EM>f. harris, "Multirate Signal Processing for Communication
     *       Systems," Upper Saddle River, NJ: Prentice Hall, Inc. 2004.</EM></B>
     *
     * When dealing with oversampling, the above book is still a good
     * reference along with this paper:
     *
     *    <B><EM>E. Venosa, X. Chen, and fred harris, “Polyphase analysis
     *       filter bank down-converts unequal channel bandwidths with
     *       arbitrary center frequencies - design I,” in SDR’10-WinnComm,
     *       2010.</EM></B>
     */
    class FILTER_API pfb_channelizer_ccf : virtual public block
    {
    public:
      // gr::filter::pfb_channelizer_ccf::sptr
      typedef boost::shared_ptr<pfb_channelizer_ccf> sptr;

      /*!
       * 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
       *                                rateis 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.
       */
      static sptr make(unsigned int numchans,
                                  const std::vector<float> &taps,
                                  float oversample_rate);

      /*!
       * Resets the filterbank's filter taps with the new prototype filter
       * \param taps (vector/list of floats) The prototype filter to populate the filterbank.
       */
      virtual void set_taps(const std::vector<float> &taps) = 0;

      /*!
       * Print all of the filterbank taps to screen.
       */
      virtual void print_taps() = 0;

      /*!
       * Return a vector<vector<>> of the filterbank taps
       */
      virtual std::vector<std::vector<float> > taps() const = 0;

      /*!
       * Set the channel map. Channels are numbers as:
       *
       *     N/2+1 | ... | N-1 | 0 | 1 |  2 | ... | N/2
       *    <------------------- 0 -------------------->
       *                        freq
       *
       * So output stream 0 comes from channel 0, etc. Setting a new
       * channel map allows the user to specify which channel in frequency
       * he/she wants to got to which output stream.
       *
       * The map should have the same number of elements as the number
       * of output connections from the block. The minimum value of
       * the map is 0 (for the 0th channel) and the maximum number is
       * N-1 where N is the number of channels.
       *
       * We specify M as the number of output connections made where M
       * <= N, so only M out of N channels are driven to an output
       * stream. The number of items in the channel map should be at
       * least M long. If there are more channels specified, any value
       * in the map over M-1 will be ignored. If the size of the map
       * is less than M the behavior is unknown (we don't wish to
       * check every entry into the work function).
       *
       * This means that if the channelizer is splitting the signal up
       * into N channels but only M channels are specified in the map
       * (where M <= N), then M output streams must be connected and
       * the map and the channel numbers used must be less than
       * N-1. Output channel number can be reused, too. By default,
       * the map is [0...M-1] with M = N.
       */
      virtual void set_channel_map(const std::vector<int> &map) = 0;

      /*!
       * Gets the current channel map.
       */
      virtual std::vector<int> channel_map() const = 0;
    };

  } /* namespace filter */
} /* namespace gr */

#endif /* INCLUDED_FILTER_PFB_CHANNELIZER_CCF_H */