def __init__(self, fg, audio_rate, quad_rate, tau=75e-6, max_dev=5e3):
"""
Narrow Band FM Receiver.
Takes a single complex baseband input stream and produces a single
float output stream of audio sample in the range [-1, +1].
@param fg: flow graph
@param audio_rate: sample rate of audio stream, >= 16k
@type audio_rate: integer
@param quad_rate: sample rate of output stream
@type quad_rate: integer
@param tau: preemphasis time constant (default 75e-6)
@type tau: float
@param max_dev: maximum deviation in Hz (default 5e3)
@type max_dev: float
quad_rate must be an integer multiple of audio_rate.
Exported sub-blocks (attributes):
squelch
quad_demod
deemph
audio_filter
"""
# FIXME audio_rate and quad_rate ought to be exact rationals
audio_rate = int(audio_rate)
quad_rate = int(quad_rate)
if quad_rate % audio_rate != 0:
raise ValueError, "quad_rate is not an integer multiple of audio_rate"
squelch_threshold = 20 # dB
#self.squelch = gr.simple_squelch_cc(squelch_threshold, 0.001)
# FM Demodulator input: complex; output: float
k = quad_rate/(2*math.pi*max_dev)
self.quad_demod = gr.quadrature_demod_cf(k)
# FM Deemphasis IIR filter
self.deemph = fm_deemph (fg, quad_rate, tau=tau)
# compute FIR taps for audio filter
audio_decim = quad_rate // audio_rate
audio_taps = gr.firdes.low_pass (1.0, # gain
quad_rate, # sampling rate
4.5e3, # Audio LPF cutoff
2.5e3, # Transition band
gr.firdes.WIN_HAMMING) # filter type
print "len(audio_taps) =", len(audio_taps)
# Decimating audio filter
# input: float; output: float; taps: float
self.audio_filter = gr.fir_filter_fff(audio_decim, audio_taps)
fg.connect(self.quad_demod, self.deemph, self.audio_filter)
gr.hier_block.__init__(self, fg, self.quad_demod, self.audio_filter)
def __init__(self, fg, channel_rate, audio_decim, deviation, audio_pass, audio_stop, gain=1.0, tau=75e-6):
"""
# Equalizer for ~100 us delay
delay = 100e-6
num_taps = int(channel_rate*delay)
mu = 1e-4/num_taps
print "CMA: delay =", delay, "n =", num_taps, "mu =", mu
CMA = gr.cma_equalizer_cc(num_taps, 1.0, mu)
"""
k = channel_rate / (2 * pi * deviation)
QUAD = gr.quadrature_demod_cf(k)
audio_taps = optfir.low_pass(
gain, # Filter gain
channel_rate, # Sample rate
audio_pass, # Audio passband
audio_stop, # Audio stopband
0.1, # Passband ripple
60,
) # Stopband attenuation
LPF = gr.fir_filter_fff(audio_decim, audio_taps)
if tau is not None:
DEEMPH = fm_deemph(fg, channel_rate, tau)
fg.connect(QUAD, DEEMPH, LPF)
else:
fg.connect(QUAD, LPF)
gr.hier_block.__init__(self, fg, QUAD, LPF)
def __init__ (self, fg, quad_rate, audio_decimation):
"""
Hierarchical block for demodulating a broadcast FM signal.
The input is the downconverted complex baseband signal (gr_complex).
The output is the demodulated audio (float).
@param fg: flow graph.
@type fg: flow graph
@param quad_rate: input sample rate of complex baseband input.
@type quad_rate: float
@param audio_decimation: how much to decimate quad_rate to get to audio.
@type audio_decimation: integer
"""
volume = 20.
max_dev = 75e3
fm_demod_gain = quad_rate/(2*math.pi*max_dev)
audio_rate = quad_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
self.fm_demod = gr.quadrature_demod_cf (fm_demod_gain)
# input: float; output: float
self.deemph = fm_deemph (fg, audio_rate)
# compute FIR filter taps for audio filter
width_of_transition_band = audio_rate / 32
audio_coeffs = gr.firdes.low_pass (1.0, # gain
quad_rate, # sampling rate
audio_rate/2 - width_of_transition_band,
width_of_transition_band,
gr.firdes.WIN_HAMMING)
# input: float; output: float
self.audio_filter = gr.fir_filter_fff (audio_decimation, audio_coeffs)
fg.connect (self.fm_demod, self.audio_filter, self.deemph)
gr.hier_block.__init__(self,
fg,
self.fm_demod, # head of the pipeline
self.deemph) # tail of the pipeline
def __init__ (self, fg, 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.
@param fg: flow graph.
@type fg: flow graph
@param demod_rate: input sample rate of complex baseband input.
@type demod_rate: float
@param audio_decimation: how much to decimate demod_rate to get to audio.
@type audio_decimation: integer
"""
bandwidth = 200e3
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
alpha = 0.25*bandwidth * math.pi / demod_rate
beta = alpha * alpha / 4.0
max_freq = 2.0*math.pi*100e3/demod_rate
self.fm_demod = gr.pll_freqdet_cf (alpha,beta,max_freq,-max_freq)
# input: float; output: float
self.deemph_Left = fm_deemph (fg, audio_rate)
self.deemph_Right = fm_deemph (fg, audio_rate)
# compute FIR filter taps for audio filter
width_of_transition_band = audio_rate / 32
audio_coeffs = gr.firdes.low_pass (1.0 , # gain
demod_rate, # sampling rate
15000 ,
width_of_transition_band,
gr.firdes.WIN_HAMMING)
# input: float; output: float
self.audio_filter = gr.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 = gr.firdes.complex_band_pass(10.0,
demod_rate,
-19020,
-18980,
width_of_transition_band,
gr.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 = gr.firdes.complex_band_pass(20.0,
demod_rate,
38000-15000/2,
38000+15000/2,
width_of_transition_band,
gr.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 = gr.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 = gr.multiply_cc();
# Pick off the rds signal
stereo_rds_filter_coeffs = gr.firdes.complex_band_pass(30.0,
demod_rate,
57000 - 1500,
57000 + 1500,
width_of_transition_band,
gr.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 = gr.fir_filter_fcc(audio_decimation, stereo_carrier_filter_coeffs)
self.rds_signal_filter = gr.fir_filter_fcc(audio_decimation, stereo_rds_filter_coeffs)
self.rds_carrier_generator = gr.multiply_cc();
self.rds_signal_generator = gr.multiply_cc();
self_rds_signal_processor = gr.null_sink(gr.sizeof_gr_complex);
alpha = 5 * 0.25 * math.pi / (audio_rate)
beta = alpha * alpha / 4.0
max_freq = -2.0*math.pi*18990/audio_rate;
min_freq = -2.0*math.pi*19010/audio_rate;
self.stereo_carrier_pll_recovery = gr.pll_refout_cc(alpha,beta,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 = gr.multiply_cc();
# pick off the real component of the basebanded L-R signal. The imaginary SHOULD be zero
self.LmR_real = gr.complex_to_real();
self.Make_Left = gr.add_ff();
self.Make_Right = gr.sub_ff();
self.stereo_dsbsc_filter = gr.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
fg.connect (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
fg.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)
fg.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
fg.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
fg.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
fg.connect (self.LmR_real,(self.Make_Right,1))
# Make rds carrier by taking the squared pilot tone and multiplying by pilot tone
fg.connect (self.stereo_basebander,(self.rds_carrier_generator,0))
fg.connect (self.stereo_carrier_pll_recovery,(self.rds_carrier_generator,1))
# take signal, filter off rds, send into mixer 0 channel
fg.connect (self.fm_demod,self.rds_signal_filter,(self.rds_signal_generator,0))
# take rds_carrier_generator output and send into mixer 1 channel
fg.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
fg.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
fg.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
fg.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
# kludge the signals into a stereo channel
kludge = gr.kludge_copy(gr.sizeof_float)
fg.connect(self.Make_Left , self.deemph_Left, (kludge, 0))
fg.connect(self.Make_Right, self.deemph_Right, (kludge, 1))
#send it to the audio system
gr.hier_block.__init__(self,
fg,
self.fm_demod, # head of the pipeline
kludge) # tail of the pipeline
else:
fg.connect (self.fm_demod, self.audio_filter)
gr.hier_block.__init__(self,
fg,
self.fm_demod, # head of the pipeline
self.audio_filter) # tail of the pipeline