diff options
Diffstat (limited to 'gr-digital')
| -rw-r--r-- | gr-digital/lib/digital_fll_band_edge_cc.h | 115 |
1 files changed, 65 insertions, 50 deletions
diff --git a/gr-digital/lib/digital_fll_band_edge_cc.h b/gr-digital/lib/digital_fll_band_edge_cc.h index 496289d90..fbd59881b 100644 --- a/gr-digital/lib/digital_fll_band_edge_cc.h +++ b/gr-digital/lib/digital_fll_band_edge_cc.h @@ -39,35 +39,47 @@ digital_fll_band_edge_cc_sptr digital_make_fll_band_edge_cc (float samps_per_sym * * \ingroup general * - * The frequency lock loop derives a band-edge filter that covers the upper and lower bandwidths - * of a digitally-modulated signal. The bandwidth range is determined by the excess bandwidth - * (e.g., rolloff factor) of the modulated signal. The placement in frequency of the band-edges - * is determined by the oversampling ratio (number of samples per symbol) and the excess bandwidth. - * The size of the filters should be fairly large so as to average over a number of symbols. + * The frequency lock loop derives a band-edge filter that covers the + * upper and lower bandwidths of a digitally-modulated signal. The + * bandwidth range is determined by the excess bandwidth (e.g., + * rolloff factor) of the modulated signal. The placement in frequency + * of the band-edges is determined by the oversampling ratio (number + * of samples per symbol) and the excess bandwidth. The size of the + * filters should be fairly large so as to average over a number of + * symbols. * - * The FLL works by filtering the upper and lower band edges into x_u(t) and x_l(t), respectively. - * These are combined to form cc(t) = x_u(t) + x_l(t) and ss(t) = x_u(t) - x_l(t). Combining - * these to form the signal e(t) = Re{cc(t) \\times ss(t)^*} (where ^* is the complex conjugate) - * provides an error signal at the DC term that is directly proportional to the carrier frequency. - * We then make a second-order loop using the error signal that is the running average of e(t). + * The FLL works by filtering the upper and lower band edges into + * x_u(t) and x_l(t), respectively. These are combined to form cc(t) + * = x_u(t) + x_l(t) and ss(t) = x_u(t) - x_l(t). Combining these to + * form the signal e(t) = Re{cc(t) \\times ss(t)^*} (where ^* is the + * complex conjugate) provides an error signal at the DC term that is + * directly proportional to the carrier frequency. We then make a + * second-order loop using the error signal that is the running + * average of e(t). * - * In practice, the above equation can be simplified by just comparing the absolute value squared - * of the output of both filters: abs(x_l(t))^2 - abs(x_u(t))^2 = norm(x_l(t)) - norm(x_u(t)). + * In practice, the above equation can be simplified by just comparing + * the absolute value squared of the output of both filters: + * abs(x_l(t))^2 - abs(x_u(t))^2 = norm(x_l(t)) - norm(x_u(t)). * - * In theory, the band-edge filter is the derivative of the matched filter in frequency, - * (H_be(f) = \\frac{H(f)}{df}. In practice, this comes down to a quarter sine wave at the point - * of the matched filter's rolloff (if it's a raised-cosine, the derivative of a cosine is a sine). - * Extend this sine by another quarter wave to make a half wave around the band-edges is equivalent - * in time to the sum of two sinc functions. The baseband filter fot the band edges is therefore - * derived from this sum of sincs. The band edge filters are then just the baseband signal - * modulated to the correct place in frequency. All of these calculations are done in the + * In theory, the band-edge filter is the derivative of the matched + * filter in frequency, (H_be(f) = \\frac{H(f)}{df}. In practice, this + * comes down to a quarter sine wave at the point of the matched + * filter's rolloff (if it's a raised-cosine, the derivative of a + * cosine is a sine). Extend this sine by another quarter wave to + * make a half wave around the band-edges is equivalent in time to the + * sum of two sinc functions. The baseband filter fot the band edges + * is therefore derived from this sum of sincs. The band edge filters + * are then just the baseband signal modulated to the correct place in + * frequency. All of these calculations are done in the * 'design_filter' function. * - * Note: We use FIR filters here because the filters have to have a flat phase response over the - * entire frequency range to allow their comparisons to be valid. + * Note: We use FIR filters here because the filters have to have a + * flat phase response over the entire frequency range to allow their + * comparisons to be valid. * - * It is very important that the band edge filters be the derivatives of the pulse shaping filter, - * and that they be linear phase. Otherwise, the variance of the error will be very large. + * It is very important that the band edge filters be the derivatives + * of the pulse shaping filter, and that they be linear + * phase. Otherwise, the variance of the error will be very large. * */ @@ -155,12 +167,12 @@ public: * \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. + * 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(). + * 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 * @@ -172,8 +184,8 @@ public: * * Set's the loop filter's alpha gain parameter. * - * This value should really only be set by adjusting the loop bandwidth - * and damping factor. + * This value should really only be set by adjusting the loop + * bandwidth and damping factor. * * \param alpha (float) new alpha gain * @@ -185,8 +197,8 @@ public: * * Set's the loop filter's beta gain parameter. * - * This value should really only be set by adjusting the loop bandwidth - * and damping factor. + * This value should really only be set by adjusting the loop + * bandwidth and damping factor. * * \param beta (float) new beta gain * @@ -196,8 +208,9 @@ public: /*! * \brief Set the number of samples per symbol * - * Set's the number of samples per symbol the system should use. This value - * is uesd to calculate the filter taps and will force a recalculation. + * Set's the number of samples per symbol the system should + * use. This value is uesd to calculate the filter taps and will + * force a recalculation. * * \param sps (float) new samples per symbol * @@ -207,13 +220,14 @@ public: /*! * \brief Set the rolloff factor of the shaping filter * - * This sets the rolloff factor that is used in the pulse shaping filter - * and is used to calculate the filter taps. Changing this will force a - * recalculation of the filter taps. + * This sets the rolloff factor that is used in the pulse shaping + * filter and is used to calculate the filter taps. Changing this + * will force a recalculation of the filter taps. * - * This should be the same value that is used in the transmitter's pulse - * shaping filter. It must be between 0 and 1 and is usually between - * 0.2 and 0.5 (where 0.22 and 0.35 are commonly used values). + * This should be the same value that is used in the transmitter's + * pulse shaping filter. It must be between 0 and 1 and is usually + * between 0.2 and 0.5 (where 0.22 and 0.35 are commonly used + * values). * * \param rolloff (float) new shaping filter rolloff factor [0,1] * @@ -223,13 +237,14 @@ public: /*! * \brief Set the number of taps in the filter * - * This sets the number of taps in the band-edge filters. Setting this will - * force a recalculation of the filter taps. + * This sets the number of taps in the band-edge filters. Setting + * this will force a recalculation of the filter taps. * - * This should be about the same number of taps used in the transmitter's - * shaping filter and also not very large. A large number of taps will - * result in a large delay between input and frequency estimation, and - * so will not be as accurate. Between 30 and 70 taps is usual. + * This should be about the same number of taps used in the + * transmitter's shaping filter and also not very large. A large + * number of taps will result in a large delay between input and + * frequency estimation, and so will not be as accurate. Between 30 + * and 70 taps is usual. * * \param filter_size (float) number of taps in the filters * @@ -240,8 +255,8 @@ public: * \brief Set the FLL's frequency. * * Set's the FLL's frequency. While this is normally updated by the - * inner loop of the algorithm, it could be useful to manually initialize, - * set, or reset this under certain circumstances. + * inner loop of the algorithm, it could be useful to manually + * initialize, set, or reset this under certain circumstances. * * \param freq (float) new frequency * @@ -252,8 +267,8 @@ public: * \brief Set the FLL's phase. * * Set's the FLL's phase. While this is normally updated by the - * inner loop of the algorithm, it could be useful to manually initialize, - * set, or reset this under certain circumstances. + * inner loop of the algorithm, it could be useful to manually + * initialize, set, or reset this under certain circumstances. * * \param phase (float) new phase * |