diff options
Diffstat (limited to 'gr-digital/include/digital_pfb_clock_sync_ccf.h')
| -rw-r--r-- | gr-digital/include/digital_pfb_clock_sync_ccf.h | 376 |
1 files changed, 376 insertions, 0 deletions
diff --git a/gr-digital/include/digital_pfb_clock_sync_ccf.h b/gr-digital/include/digital_pfb_clock_sync_ccf.h new file mode 100644 index 000000000..f7996d175 --- /dev/null +++ b/gr-digital/include/digital_pfb_clock_sync_ccf.h @@ -0,0 +1,376 @@ +/* -*- 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_CCF_H +#define INCLUDED_DIGITAL_PFB_CLOCK_SYNC_CCF_H + +#include <digital_api.h> +#include <gr_block.h> + +class digital_pfb_clock_sync_ccf; +typedef boost::shared_ptr<digital_pfb_clock_sync_ccf> digital_pfb_clock_sync_ccf_sptr; +DIGITAL_API digital_pfb_clock_sync_ccf_sptr +digital_make_pfb_clock_sync_ccf(double sps, float loop_bw, + const std::vector<float> &taps, + unsigned int filter_size=32, + float init_phase=0, + float max_rate_deviation=1.5, + int osps=1); + +class gr_fir_ccf; + +/*! + * \class digital_pfb_clock_sync_ccf + * + * \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 GR_CORE_API digital_pfb_clock_sync_ccf : public gr_block +{ + private: + /*! + * Build the polyphase filterbank timing synchronizer. + * \param sps (double) The number of samples per symbol in the incoming signal + * \param loop_bw (float) The bandwidth of the control loop; set's alpha and beta. + * \param taps (vector<int>) 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_ccf_sptr + digital_make_pfb_clock_sync_ccf(double sps, float loop_bw, + const std::vector<float> &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<gr_fir_ccf*> d_filters; + std::vector<gr_fir_ccf*> d_diff_filters; + std::vector< std::vector<float> > d_taps; + std::vector< std::vector<float> > 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_ccf(double sps, float loop_bw, + const std::vector<float> &taps, + unsigned int filter_size, + float init_phase, + float max_rate_deviation, + int osps); + + void create_diff_taps(const std::vector<float> &newtaps, + std::vector<float> &difftaps); + +public: + ~digital_pfb_clock_sync_ccf(); + + /*! \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<float> &taps, + std::vector< std::vector<float> > &ourtaps, + std::vector<gr_fir_ccf*> &ourfilter); + + /*! + * Returns all of the taps of the matched filter + */ + std::vector< std::vector<float> > get_taps(); + + /*! + * Returns all of the taps of the derivative filter + */ + std::vector< std::vector<float> > get_diff_taps(); + + /*! + * Returns the taps of the matched filter for a particular channel + */ + std::vector<float> get_channel_taps(int channel); + + /*! + * Returns the taps in the derivative filter for a particular channel + */ + std::vector<float> 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 |