#
# Copyright 2005,2006,2012-2013 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.
#
from gnuradio import gr
from gnuradio import blocks
from gnuradio import filter
from fm_emph import fm_deemph
import math
try:
from gnuradio import analog
except ImportError:
import analog_swig as analog
class wfm_rcv_pll(gr.hier_block2):
def __init__(self, demod_rate, audio_decimation):
"""
Hierarchical block for demodulating a broadcast FM signal.
The input is the downconverted complex baseband signal (gr_complex).
The output is two streams of the demodulated audio (float) 0=Left, 1=Right.
Args:
demod_rate: input sample rate of complex baseband input. (float)
audio_decimation: how much to decimate demod_rate to get to audio. (integer)
"""
gr.hier_block2.__init__(self, "wfm_rcv_pll",
gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature
gr.io_signature(2, 2, gr.sizeof_float)) # Output signature
bandwidth = 250e3
audio_rate = demod_rate / audio_decimation
# We assign to self so that outsiders can grab the demodulator
# if they need to. E.g., to plot its output.
#
# input: complex; output: float
loop_bw = 2*math.pi/100.0
max_freq = 2.0*math.pi*90e3/demod_rate
self.fm_demod = analog.pll_freqdet_cf(loop_bw, max_freq,-max_freq)
# input: float; output: float
self.deemph_Left = fm_deemph(audio_rate)
self.deemph_Right = fm_deemph(audio_rate)
# compute FIR filter taps for audio filter
width_of_transition_band = audio_rate / 32
audio_coeffs = filter.firdes.low_pass(1.0 , # gain
demod_rate, # sampling rate
15000 ,
width_of_transition_band,
filter.firdes.WIN_HAMMING)
# input: float; output: float
self.audio_filter = filter.fir_filter_fff(audio_decimation, audio_coeffs)
if 1:
# Pick off the stereo carrier/2 with this filter. It attenuated 10 dB so apply 10 dB gain
# We pick off the negative frequency half because we want to base band by it!
## NOTE THIS WAS HACKED TO OFFSET INSERTION LOSS DUE TO DEEMPHASIS
stereo_carrier_filter_coeffs = \
filter.firdes.complex_band_pass(10.0,
demod_rate,
-19020,
-18980,
width_of_transition_band,
filter.firdes.WIN_HAMMING)
#print "len stereo carrier filter = ",len(stereo_carrier_filter_coeffs)
#print "stereo carrier filter ", stereo_carrier_filter_coeffs
#print "width of transition band = ",width_of_transition_band, " audio rate = ", audio_rate
# Pick off the double side band suppressed carrier Left-Right audio. It is attenuated 10 dB so apply 10 dB gain
stereo_dsbsc_filter_coeffs = \
filter.firdes.complex_band_pass(20.0,
demod_rate,
38000-15000/2,
38000+15000/2,
width_of_transition_band,
filter.firdes.WIN_HAMMING)
#print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs)
#print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs
# construct overlap add filter system from coefficients for stereo carrier
self.stereo_carrier_filter = \
filter.fir_filter_fcc(audio_decimation, stereo_carrier_filter_coeffs)
# carrier is twice the picked off carrier so arrange to do a commplex multiply
self.stereo_carrier_generator = blocks.multiply_cc();
# Pick off the rds signal
stereo_rds_filter_coeffs = \
filter.firdes.complex_band_pass(30.0,
demod_rate,
57000 - 1500,
57000 + 1500,
width_of_transition_band,
filter.firdes.WIN_HAMMING)
#print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs)
#print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs
# construct overlap add filter system from coefficients for stereo carrier
self.rds_signal_filter = \
filter.fir_filter_fcc(audio_decimation, stereo_rds_filter_coeffs)
self.rds_carrier_generator = blocks.multiply_cc();
self.rds_signal_generator = blocks.multiply_cc();
self_rds_signal_processor = blocks.null_sink(gr.sizeof_gr_complex);
loop_bw = 2*math.pi/100.0
max_freq = -2.0*math.pi*18990/audio_rate;
min_freq = -2.0*math.pi*19010/audio_rate;
self.stereo_carrier_pll_recovery = \
analog.pll_refout_cc(loop_bw, max_freq, min_freq);
#self.stereo_carrier_pll_recovery.squelch_enable(False) #pll_refout does not have squelch yet, so disabled for now
# set up mixer (multiplier) to get the L-R signal at baseband
self.stereo_basebander = blocks.multiply_cc();
# pick off the real component of the basebanded L-R signal. The imaginary SHOULD be zero
self.LmR_real = blocks.complex_to_real();
self.Make_Left = blocks.add_ff();
self.Make_Right = blocks.sub_ff();
self.stereo_dsbsc_filter = \
filter.fir_filter_fcc(audio_decimation, stereo_dsbsc_filter_coeffs)
if 1:
# send the real signal to complex filter to pick off the carrier and then to one side of a multiplier
self.connect(self, self.fm_demod, self.stereo_carrier_filter,
self.stereo_carrier_pll_recovery, (self.stereo_carrier_generator,0))
# send the already filtered carrier to the otherside of the carrier
self.connect(self.stereo_carrier_pll_recovery, (self.stereo_carrier_generator,1))
# the resulting signal from this multiplier is the carrier with correct phase but at -38000 Hz.
# send the new carrier to one side of the mixer (multiplier)
self.connect(self.stereo_carrier_generator, (self.stereo_basebander,0))
# send the demphasized audio to the DSBSC pick off filter, the complex
# DSBSC signal at +38000 Hz is sent to the other side of the mixer/multiplier
self.connect(self.fm_demod,self.stereo_dsbsc_filter, (self.stereo_basebander,1))
# the result is BASEBANDED DSBSC with phase zero!
# Pick off the real part since the imaginary is theoretically zero and then to one side of a summer
self.connect(self.stereo_basebander, self.LmR_real, (self.Make_Left,0))
#take the same real part of the DSBSC baseband signal and send it to negative side of a subtracter
self.connect(self.LmR_real,(self.Make_Right,1))
# Make rds carrier by taking the squared pilot tone and multiplying by pilot tone
self.connect(self.stereo_basebander,(self.rds_carrier_generator,0))
self.connect(self.stereo_carrier_pll_recovery,(self.rds_carrier_generator,1))
# take signal, filter off rds, send into mixer 0 channel
self.connect(self.fm_demod,self.rds_signal_filter,(self.rds_signal_generator,0))
# take rds_carrier_generator output and send into mixer 1 channel
self.connect(self.rds_carrier_generator,(self.rds_signal_generator,1))
# send basebanded rds signal and send into "processor" which for now is a null sink
self.connect(self.rds_signal_generator,self_rds_signal_processor)
if 1:
# pick off the audio, L+R that is what we used to have and send it to the summer
self.connect(self.fm_demod, self.audio_filter, (self.Make_Left, 1))
# take the picked off L+R audio and send it to the PLUS side of the subtractor
self.connect(self.audio_filter,(self.Make_Right, 0))
# The result of Make_Left gets (L+R) + (L-R) and results in 2*L
# The result of Make_Right gets (L+R) - (L-R) and results in 2*R
self.connect(self.Make_Left , self.deemph_Left, (self, 0))
self.connect(self.Make_Right, self.deemph_Right, (self, 1))
# NOTE: mono support will require variable number of outputs in hier_block2s
# See ticket:174 in Trac database
#else:
# self.connect (self.fm_demod, self.audio_filter, self)