I reached essentially the same conclusions that Chuck reached afew weeks back on
SSB Phasing receiver, the code of which is given below. The program seem to be
a son't care on the sideband and demodulates lsb even if we ask it to demodulate
usb. Can someone throw light on it? I derived all the math with the preenvelope
calculations etc and verified it. It all seem to be right.
It seem to me that hilbert transform is goofing up somewhere. The next step may
be to verify whether the coefficients of hilbert filter look alright or not.
#!/usr/bin/python
# SSB receiver using I/Q synchronous detection
# Donald ducking is due to difference in freq b/n carrier and LO.
# copyright (c) 2005, FSF under the GNU GPL v2 or later
# Author: Ramakrishnan Muthukrishnan <[EMAIL PROTECTED]>
#
# Phasing SSB receiver based on block diagram from the article
# "High Performance, Single-Signal Direct Conversion Receivers"
# by Rick Campbell, KK7B, QST Jan 1993
# Program usage with the OTA capture samples from KD7LMO:
# ssb_phasing.py offset bfo ssb_lsb_256k_complex.dat {l|L|u|U}
# eg: ssb_phasing.py 50000 3100 ssb_lsb_256k_complex.dat l
from gnuradio import gr
from gnuradio import audio
import sys
import os
def build_graph (filename, if_freq, side_band):
sampling_freq = 256e3
bandwidth = 10e3 # 20khz
trans_bw = 1e3
audio_rate = 48e3 # 32khz
decim = long (sampling_freq / audio_rate)
rf_lo = if_freq
# file is our source.
src = gr.file_source (gr.sizeof_gr_complex, filename, 1)
# LO
lo_i = gr.sig_source_f (sampling_freq,gr.GR_SIN_WAVE,rf_lo,1.0,0)
lo_q = gr.sig_source_f (sampling_freq,gr.GR_COS_WAVE,rf_lo,1.0,0)
# mixers
mix_i = gr.multiply_ff ()
mix_q = gr.multiply_ff ()
# hilbert coefficients for N = 49 taps
# perform hilbert transform and delay. Output is gr_complex
n_taps = 79
hilb_filter = gr.firdes_hilbert (n_taps, gr.firdes.WIN_HAMMING)
hilbert = gr.fir_filter_fff (1, hilb_filter)
delay_taps = []
for i in range((n_taps-1)/2):
delay_taps.append (0)
delay_taps[((n_taps-1)/2) - 1] = 1
delay = gr.fir_filter_fff (1, delay_taps)
# output of the above block is complex. Split into floats
split1 = gr.complex_to_float ()
# create LPFs
lpf_coeffs = gr.firdes.low_pass ( \
1.0,
audio_rate,
5e3,
500,
gr.firdes.WIN_HAMMING)
lpf1 = gr.fir_filter_fff ( \
decim,
lpf_coeffs)
lpf2 = gr.fir_filter_fff ( \
decim,
lpf_coeffs)
# adder
if side_band == 'L' or side_band == 'l':
sum = gr.add_ff ()
else:
sum = gr.sub_ff ()
adder = gr.add_ff ()
sub = gr.sub_ff ()
gain = gr.multiply_const_ff(.00002)
# audio output
audio_sink = audio.sink (int (audio_rate))
# add required blocks created. Now build the flow graph
fg = gr.flow_graph ()
fg.connect (src, split1)
fg.connect ((split1,0), (mix_i,0))
fg.connect (lo_i, (mix_i,1))
fg.connect ((split1,1), (mix_q,0))
fg.connect (lo_q, (mix_q,1))
fg.connect (mix_i, lpf1)
fg.connect (mix_q, lpf2)
fg.connect (lpf1, delay)
fg.connect (lpf2, hilbert)
fg.connect (delay, (sum,0))
fg.connect (hilbert, (sum,1))
fg.connect (sum, gain)
fg.connect (gain, audio_sink)
return fg
def main (args):
filename = args[2]
bfo = float(args[1])
freq = float (args[0])
freq = freq + bfo
print "Tuning to freq %d" % (freq)
print "filename is %s" %(filename)
fg = build_graph (filename, freq,args[3][0])
fg.start ()
raw_input ('Press Enter to quit: ')
fg.stop ()
if __name__ == '__main__':
main (sys.argv[1:])
--
Ramakrishnan http://www.hackGNU.org/
Use Free Software -- Help stamp out Software Hoarding!
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