GNU Radio 3.7.1 C++ API
pfb_clock_sync_fff.h
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00001 /* -*- c++ -*- */
00002 /*
00003  * Copyright 2009,2010,2012 Free Software Foundation, Inc.
00004  *
00005  * This file is part of GNU Radio
00006  *
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00008  * it under the terms of the GNU General Public License as published by
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00010  * any later version.
00011  *
00012  * GNU Radio is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
00015  * GNU General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU General Public License
00018  * along with GNU Radio; see the file COPYING.  If not, write to
00019  * the Free Software Foundation, Inc., 51 Franklin Street,
00020  * Boston, MA 02110-1301, USA.
00021  */
00022 
00023 #ifndef INCLUDED_DIGITAL_PFB_CLOCK_SYNC_FFF_H
00024 #define INCLUDED_DIGITAL_PFB_CLOCK_SYNC_FFF_H
00025 
00026 #include <gnuradio/digital/api.h>
00027 #include <gnuradio/filter/fir_filter.h>
00028 #include <gnuradio/block.h>
00029 
00030 namespace gr {
00031   namespace digital {
00032 
00033     /*!
00034      * \brief Timing synchronizer using polyphase filterbanks
00035      * \ingroup synchronizers_blk
00036      *
00037      * \details
00038      * This block performs timing synchronization for PAM signals by
00039      * minimizing the derivative of the filtered signal, which in turn
00040      * maximizes the SNR and minimizes ISI.
00041      *
00042      * This approach works by setting up two filterbanks; one
00043      * filterbank contains the signal's pulse shaping matched filter
00044      * (such as a root raised cosine filter), where each branch of the
00045      * filterbank contains a different phase of the filter.  The
00046      * second filterbank contains the derivatives of the filters in
00047      * the first filterbank. Thinking of this in the time domain, the
00048      * first filterbank contains filters that have a sinc shape to
00049      * them. We want to align the output signal to be sampled at
00050      * exactly the peak of the sinc shape. The derivative of the sinc
00051      * contains a zero at the maximum point of the sinc (sinc(0) = 1,
00052      * sinc(0)' = 0).  Furthermore, the region around the zero point
00053      * is relatively linear. We make use of this fact to generate the
00054      * error signal.
00055      *
00056      * If the signal out of the derivative filters is d_i[n] for the
00057      * ith filter, and the output of the matched filter is x_i[n], we
00058      * calculate the error as: e[n] = (Re{x_i[n]} * Re{d_i[n]} +
00059      * Im{x_i[n]} * Im{d_i[n]}) / 2.0 This equation averages the error
00060      * in the real and imaginary parts. There are two reasons we
00061      * multiply by the signal itself. First, if the symbol could be
00062      * positive or negative going, but we want the error term to
00063      * always tell us to go in the same direction depending on which
00064      * side of the zero point we are on. The sign of x_i[n] adjusts
00065      * the error term to do this. Second, the magnitude of x_i[n]
00066      * scales the error term depending on the symbol's amplitude, so
00067      * larger signals give us a stronger error term because we have
00068      * more confidence in that symbol's value.  Using the magnitude of
00069      * x_i[n] instead of just the sign is especially good for signals
00070      * with low SNR.
00071      *
00072      * The error signal, e[n], gives us a value proportional to how
00073      * far away from the zero point we are in the derivative
00074      * signal. We want to drive this value to zero, so we set up a
00075      * second order loop. We have two variables for this loop; d_k is
00076      * the filter number in the filterbank we are on and d_rate is the
00077      * rate which we travel through the filters in the steady
00078      * state. That is, due to the natural clock differences between
00079      * the transmitter and receiver, d_rate represents that difference
00080      * and would traverse the filter phase paths to keep the receiver
00081      * locked. Thinking of this as a second-order PLL, the d_rate is
00082      * the frequency and d_k is the phase. So we update d_rate and d_k
00083      * using the standard loop equations based on two error signals,
00084      * d_alpha and d_beta.  We have these two values set based on each
00085      * other for a critically damped system, so in the block
00086      * constructor, we just ask for "gain," which is d_alpha while
00087      * d_beta is equal to (gain^2)/4.
00088      *
00089      * The block's parameters are:
00090      *
00091      * \li \p sps: The clock sync block needs to know the number of
00092      * samples per symbol, because it defaults to return a single
00093      * point representing the symbol. The sps can be any positive real
00094      * number and does not need to be an integer.
00095      *
00096      * \li \p loop_bw: The loop bandwidth is used to set the gain of
00097      * the inner control loop (see:
00098      * http://gnuradio.squarespace.com/blog/2011/8/13/control-loop-gain-values.html).
00099      * This should be set small (a value of around 2pi/100 is
00100      * suggested in that blog post as the step size for the number of
00101      * radians around the unit circle to move relative to the error).
00102      *
00103      * \li \p taps: One of the most important parameters for this
00104      * block is the taps of the filter. One of the benefits of this
00105      * algorithm is that you can put the matched filter in here as the
00106      * taps, so you get both the matched filter and sample timing
00107      * correction in one go. So create your normal matched filter. For
00108      * a typical digital modulation, this is a root raised cosine
00109      * filter. The number of taps of this filter is based on how long
00110      * you expect the channel to be; that is, how many symbols do you
00111      * want to combine to get the current symbols energy back (there's
00112      * probably a better way of stating that). It's usually 5 to 10 or
00113      * so. That gives you your filter, but now we need to think about
00114      * it as a filter with different phase profiles in each filter. So
00115      * take this number of taps and multiply it by the number of
00116      * filters. This is the number you would use to create your
00117      * prototype filter. When you use this in the PFB filerbank, it
00118      * segments these taps into the filterbanks in such a way that
00119      * each bank now represents the filter at different phases,
00120      * equally spaced at 2pi/N, where N is the number of filters.
00121      *
00122      * \li \p filter_size (default=32): The number of filters can also
00123      * be set and defaults to 32. With 32 filters, you get a good
00124      * enough resolution in the phase to produce very small, almost
00125      * unnoticeable, ISI.  Going to 64 filters can reduce this more,
00126      * but after that there is very little gained for the extra
00127      * complexity.
00128      *
00129      * \li \p init_phase (default=0): The initial phase is another
00130      * settable parameter and refers to the filter path the algorithm
00131      * initially looks at (i.e., d_k starts at init_phase). This value
00132      * defaults to zero, but it might be useful to start at a
00133      * different phase offset, such as the mid-point of the filters.
00134      *
00135      * \li \p max_rate_deviation (default=1.5): The next parameter is
00136      * the max_rate_devitation, which defaults to 1.5. This is how far
00137      * we allow d_rate to swing, positive or negative, from
00138      * 0. Constraining the rate can help keep the algorithm from
00139      * walking too far away to lock during times when there is no
00140      * signal.
00141      *
00142      * \li \p osps (default=1): The osps is the number of output
00143      * samples per symbol. By default, the algorithm produces 1 sample
00144      * per symbol, sampled at the exact sample value. This osps value
00145      * was added to better work with equalizers, which do a better job
00146      * of modeling the channel if they have 2 samps/sym.
00147      */
00148     class DIGITAL_API pfb_clock_sync_fff : virtual public block
00149     {
00150     public:
00151       // gr::digital::pfb_clock_sync_fff::sptr
00152       typedef boost::shared_ptr<pfb_clock_sync_fff> sptr;
00153 
00154       /*!
00155        * Build the polyphase filterbank timing synchronizer.
00156        * \param sps (double) The number of samples per second in the incoming signal
00157        * \param gain (float) The alpha gain of the control loop; beta = (gain^2)/4 by default.
00158        * \param taps (vector<int>) The filter taps.
00159        * \param filter_size (uint) The number of filters in the filterbank (default = 32).
00160        * \param init_phase (float) The initial phase to look at, or which filter to start
00161        *                           with (default = 0).
00162        * \param max_rate_deviation (float) Distance from 0 d_rate can get (default = 1.5).
00163        * \param osps (int) The number of output samples per symbol (default=1).
00164        *
00165        */
00166       static sptr make(double sps, float gain,
00167                        const std::vector<float> &taps,
00168                        unsigned int filter_size=32,
00169                        float init_phase=0,
00170                        float max_rate_deviation=1.5,
00171                        int osps=1);
00172 
00173       /*! \brief update the system gains from omega and eta
00174        *
00175        *  This function updates the system gains based on the loop
00176        *  bandwidth and damping factor of the system.
00177        *  These two factors can be set separately through their own
00178        *  set functions.
00179        */
00180       virtual void update_gains() = 0;
00181 
00182       /*!
00183        * Resets the filterbank's filter taps with the new prototype filter
00184        */
00185       virtual void set_taps(const std::vector<float> &taps,
00186                             std::vector< std::vector<float> > &ourtaps,
00187                             std::vector<gr::filter::kernel::fir_filter_fff*> &ourfilter) = 0;
00188 
00189       /*!
00190        * Returns all of the taps of the matched filter
00191        */
00192       virtual std::vector< std::vector<float> > taps() const = 0;
00193 
00194       /*!
00195        * Returns all of the taps of the derivative filter
00196        */
00197       virtual std::vector< std::vector<float> > diff_taps() const = 0;
00198 
00199       /*!
00200        * Returns the taps of the matched filter for a particular channel
00201        */
00202       virtual std::vector<float> channel_taps(int channel) const = 0;
00203 
00204       /*!
00205        * Returns the taps in the derivative filter for a particular channel
00206        */
00207       virtual std::vector<float> diff_channel_taps(int channel) const = 0;
00208 
00209       /*!
00210        * Return the taps as a formatted string for printing
00211        */
00212       virtual std::string taps_as_string() const = 0;
00213 
00214       /*!
00215        * Return the derivative filter taps as a formatted string for printing
00216        */
00217       virtual std::string diff_taps_as_string() const = 0;
00218 
00219 
00220       /*******************************************************************
00221        SET FUNCTIONS
00222       *******************************************************************/
00223 
00224 
00225       /*!
00226        * \brief Set the loop bandwidth
00227        *
00228        * Set the loop filter's bandwidth to \p bw. This should be
00229        * between 2*pi/200 and 2*pi/100 (in rads/samp). It must also be
00230        * a positive number.
00231        *
00232        * When a new damping factor is set, the gains, alpha and beta,
00233        * of the loop are recalculated by a call to update_gains().
00234        *
00235        * \param bw    (float) new bandwidth
00236        */
00237       virtual void set_loop_bandwidth(float bw) = 0;
00238 
00239       /*!
00240        * \brief Set the loop damping factor
00241        *
00242        * Set the loop filter's damping factor to \p df. The damping
00243        * factor should be sqrt(2)/2.0 for critically damped systems.
00244        * Set it to anything else only if you know what you are
00245        * doing. It must be a number between 0 and 1.
00246        *
00247        * When a new damping factor is set, the gains, alpha and beta,
00248        * of the loop are recalculated by a call to update_gains().
00249        *
00250        * \param df    (float) new damping factor
00251        */
00252       virtual void set_damping_factor(float df) = 0;
00253 
00254       /*!
00255        * \brief Set the loop gain alpha
00256        *
00257        * Set's the loop filter's alpha gain parameter.
00258        *
00259        * This value should really only be set by adjusting the loop
00260        * bandwidth and damping factor.
00261        *
00262        * \param alpha    (float) new alpha gain
00263        */
00264       virtual void set_alpha(float alpha) = 0;
00265 
00266       /*!
00267        * \brief Set the loop gain beta
00268        *
00269        * Set's the loop filter's beta gain parameter.
00270        *
00271        * This value should really only be set by adjusting the loop
00272        * bandwidth and damping factor.
00273        *
00274        * \param beta    (float) new beta gain
00275        */
00276       virtual void set_beta(float beta) = 0;
00277 
00278       /*!
00279        * Set the maximum deviation from 0 d_rate can have
00280        */
00281       virtual void set_max_rate_deviation(float m) = 0;
00282 
00283       /*******************************************************************
00284        GET FUNCTIONS
00285       *******************************************************************/
00286 
00287       /*!
00288        * \brief Returns the loop bandwidth
00289        */
00290       virtual float loop_bandwidth() const = 0;
00291 
00292       /*!
00293        * \brief Returns the loop damping factor
00294        */
00295       virtual float damping_factor() const = 0;
00296 
00297       /*!
00298        * \brief Returns the loop gain alpha
00299        */
00300       virtual float alpha() const = 0;
00301 
00302       /*!
00303        * \brief Returns the loop gain beta
00304        */
00305       virtual float beta() const = 0;
00306 
00307       /*!
00308        * \brief Returns the current clock rate
00309        */
00310       virtual float clock_rate() const = 0;
00311     };
00312 
00313   } /* namespace digital */
00314 } /* namespace gr */
00315 
00316 #endif /* INCLUDED_DIGITAL_PFB_CLOCK_SYNC_FFF_H */