/* -*- 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_DIGITAL_PFB_CLOCK_SYNC_FFF_H #define INCLUDED_DIGITAL_PFB_CLOCK_SYNC_FFF_H #include #include class digital_pfb_clock_sync_fff; typedef boost::shared_ptr digital_pfb_clock_sync_fff_sptr; DIGITAL_API digital_pfb_clock_sync_fff_sptr digital_make_pfb_clock_sync_fff(double sps, float gain, const std::vector &taps, unsigned int filter_size=32, float init_phase=0, float max_rate_deviation=1.5, int osps=1); class gr_fir_fff; /*! * \class digital_pfb_clock_sync_fff * * \brief Timing synchronizer using polyphase filterbanks * * \ingroup filter_blk * \ingroup pfb_blk * * This block performs timing synchronization for PAM signals by * minimizing the derivative of the filtered signal, which in turn * maximizes the SNR and minimizes ISI. * * This approach works by setting up two filterbanks; one filterbank * contains the signal's pulse shaping matched filter (such as a root * raised cosine filter), where each branch of the filterbank contains * a different phase of the filter. The second filterbank contains * the derivatives of the filters in the first filterbank. Thinking of * this in the time domain, the first filterbank contains filters that * have a sinc shape to them. We want to align the output signal to be * sampled at exactly the peak of the sinc shape. The derivative of * the sinc contains a zero at the maximum point of the sinc (sinc(0) * = 1, sinc(0)' = 0). Furthermore, the region around the zero point * is relatively linear. We make use of this fact to generate the * error signal. * * If the signal out of the derivative filters is d_i[n] for the ith * filter, and the output of the matched filter is x_i[n], we * calculate the error as: e[n] = (Re{x_i[n]} * Re{d_i[n]} + * Im{x_i[n]} * Im{d_i[n]}) / 2.0 This equation averages the error in * the real and imaginary parts. There are two reasons we multiply by * the signal itself. First, if the symbol could be positive or * negative going, but we want the error term to always tell us to go * in the same direction depending on which side of the zero point we * are on. The sign of x_i[n] adjusts the error term to do * this. Second, the magnitude of x_i[n] scales the error term * depending on the symbol's amplitude, so larger signals give us a * stronger error term because we have more confidence in that * symbol's value. Using the magnitude of x_i[n] instead of just the * sign is especially good for signals with low SNR. * * The error signal, e[n], gives us a value proportional to how far * away from the zero point we are in the derivative signal. We want * to drive this value to zero, so we set up a second order loop. We * have two variables for this loop; d_k is the filter number in the * filterbank we are on and d_rate is the rate which we travel through * the filters in the steady state. That is, due to the natural clock * differences between the transmitter and receiver, d_rate represents * that difference and would traverse the filter phase paths to keep * the receiver locked. Thinking of this as a second-order PLL, the * d_rate is the frequency and d_k is the phase. So we update d_rate * and d_k using the standard loop equations based on two error * signals, d_alpha and d_beta. We have these two values set based on * each other for a critically damped system, so in the block * constructor, we just ask for "gain," which is d_alpha while d_beta * is equal to (gain^2)/4. * * The block's parameters are: * * \li \p sps: The clock sync block needs to know the number of samples per * symbol, because it defaults to return a single point representing * the symbol. The sps can be any positive real number and does not * need to be an integer. * * \li \p loop_bw: The loop bandwidth is used to set the gain of the * inner control loop (see: * http://gnuradio.squarespace.com/blog/2011/8/13/control-loop-gain-values.html). * This should be set small (a value of around 2pi/100 is suggested in * that blog post as the step size for the number of radians around * the unit circle to move relative to the error). * * \li \p taps: One of the most important parameters for this block is * the taps of the filter. One of the benefits of this algorithm is * that you can put the matched filter in here as the taps, so you get * both the matched filter and sample timing correction in one go. So * create your normal matched filter. For a typical digital * modulation, this is a root raised cosine filter. The number of taps * of this filter is based on how long you expect the channel to be; * that is, how many symbols do you want to combine to get the current * symbols energy back (there's probably a better way of stating * that). It's usually 5 to 10 or so. That gives you your filter, but * now we need to think about it as a filter with different phase * profiles in each filter. So take this number of taps and multiply * it by the number of filters. This is the number you would use to * create your prototype filter. When you use this in the PFB * filerbank, it segments these taps into the filterbanks in such a * way that each bank now represents the filter at different phases, * equally spaced at 2pi/N, where N is the number of filters. * * \li \p filter_size (default=32): The number of filters can also be * set and defaults to 32. With 32 filters, you get a good enough * resolution in the phase to produce very small, almost unnoticeable, * ISI. Going to 64 filters can reduce this more, but after that * there is very little gained for the extra complexity. * * \li \p init_phase (default=0): The initial phase is another * settable parameter and refers to the filter path the algorithm * initially looks at (i.e., d_k starts at init_phase). This value * defaults to zero, but it might be useful to start at a different * phase offset, such as the mid-point of the filters. * * \li \p max_rate_deviation (default=1.5): The next parameter is the * max_rate_devitation, which defaults to 1.5. This is how far we * allow d_rate to swing, positive or negative, from 0. Constraining * the rate can help keep the algorithm from walking too far away to * lock during times when there is no signal. * * \li \p osps (default=1): The osps is the number of output samples * per symbol. By default, the algorithm produces 1 sample per symbol, * sampled at the exact sample value. This osps value was added to * better work with equalizers, which do a better job of modeling the * channel if they have 2 samps/sym. */ class DIGITAL_API digital_pfb_clock_sync_fff : public gr_block { private: /*! * Build the polyphase filterbank timing synchronizer. * \param sps (double) The number of samples per second in the incoming signal * \param gain (float) The alpha gain of the control loop; beta = (gain^2)/4 by default. * \param taps (vector) The filter taps. * \param filter_size (uint) The number of filters in the filterbank (default = 32). * \param init_phase (float) The initial phase to look at, or which filter to start * with (default = 0). * \param max_rate_deviation (float) Distance from 0 d_rate can get (default = 1.5). * \param osps (int) The number of output samples per symbol (default=1). * */ friend DIGITAL_API digital_pfb_clock_sync_fff_sptr digital_make_pfb_clock_sync_fff(double sps, float gain, const std::vector &taps, unsigned int filter_size, float init_phase, float max_rate_deviation, int osps); bool d_updated; double d_sps; double d_sample_num; float d_loop_bw; float d_damping; float d_alpha; float d_beta; int d_nfilters; int d_taps_per_filter; std::vector d_filters; std::vector d_diff_filters; std::vector< std::vector > d_taps; std::vector< std::vector > d_dtaps; float d_k; float d_rate; float d_rate_i; float d_rate_f; float d_max_dev; int d_filtnum; int d_osps; float d_error; int d_out_idx; /*! * Build the polyphase filterbank timing synchronizer. */ digital_pfb_clock_sync_fff(double sps, float gain, const std::vector &taps, unsigned int filter_size, float init_phase, float max_rate_deviation, int osps); void create_diff_taps(const std::vector &newtaps, std::vector &difftaps); public: ~digital_pfb_clock_sync_fff (); /*! \brief update the system gains from omega and eta * * This function updates the system gains based on the loop * bandwidth and damping factor of the system. * These two factors can be set separately through their own * set functions. */ void update_gains(); /*! * Resets the filterbank's filter taps with the new prototype filter */ void set_taps(const std::vector &taps, std::vector< std::vector > &ourtaps, std::vector &ourfilter); /*! * Returns all of the taps of the matched filter */ std::vector< std::vector > get_taps(); /*! * Returns all of the taps of the derivative filter */ std::vector< std::vector > get_diff_taps(); /*! * Returns the taps of the matched filter for a particular channel */ std::vector get_channel_taps(int channel); /*! * Returns the taps in the derivative filter for a particular channel */ std::vector get_diff_channel_taps(int channel); /*! * Return the taps as a formatted string for printing */ std::string get_taps_as_string(); /*! * Return the derivative filter taps as a formatted string for printing */ std::string get_diff_taps_as_string(); /******************************************************************* SET FUNCTIONS *******************************************************************/ /*! * \brief Set the loop bandwidth * * Set the loop filter's bandwidth to \p bw. This should be between * 2*pi/200 and 2*pi/100 (in rads/samp). It must also be a positive * number. * * When a new damping factor is set, the gains, alpha and beta, of the loop * are recalculated by a call to update_gains(). * * \param bw (float) new bandwidth * */ void set_loop_bandwidth(float bw); /*! * \brief Set the loop damping factor * * Set the loop filter's damping factor to \p df. The damping factor * should be sqrt(2)/2.0 for critically damped systems. * Set it to anything else only if you know what you are doing. It must * be a number between 0 and 1. * * When a new damping factor is set, the gains, alpha and beta, of the loop * are recalculated by a call to update_gains(). * * \param df (float) new damping factor * */ void set_damping_factor(float df); /*! * \brief Set the loop gain alpha * * Set's the loop filter's alpha gain parameter. * * This value should really only be set by adjusting the loop bandwidth * and damping factor. * * \param alpha (float) new alpha gain * */ void set_alpha(float alpha); /*! * \brief Set the loop gain beta * * Set's the loop filter's beta gain parameter. * * This value should really only be set by adjusting the loop bandwidth * and damping factor. * * \param beta (float) new beta gain * */ void set_beta(float beta); /*! * Set the maximum deviation from 0 d_rate can have */ void set_max_rate_deviation(float m) { d_max_dev = m; } /******************************************************************* GET FUNCTIONS *******************************************************************/ /*! * \brief Returns the loop bandwidth */ float get_loop_bandwidth() const; /*! * \brief Returns the loop damping factor */ float get_damping_factor() const; /*! * \brief Returns the loop gain alpha */ float get_alpha() const; /*! * \brief Returns the loop gain beta */ float get_beta() const; /*! * \brief Returns the current clock rate */ float get_clock_rate() const; /******************************************************************* *******************************************************************/ bool check_topology(int ninputs, int noutputs); 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