At 12:56 PM 11/12/2005 -0500, you wrote:
Hey there,
I'm trying to create some simple transmitter programs as a basic
tutorial for myself, and I was wondering if someone would take a
quick look at my maximally simple AM transmitter program (just
trying to transmit a tone) and tell me where I've gone wrong.
Here's the program:
from gnuradio import gr
from gnuradio import audio
from gnuradio import usrp
from gnuradio.eng_option import eng_option
def build_graph ():
fg = gr.flow_graph ()
sampling_freq = 8000;
src1 = gr.sig_source_f ( sampling_freq, gr.GR_SIN_WAVE, 1090000,
4000000 )
src2 = gr.sig_source_f ( sampling_freq, gr.GR_SIN_WAVE, 600, 14000 )
dst = audio.sink (int (sampling_freq))
mixer = gr.multiply_ff()
fg.connect (src1, (mixer, 0))
fg.connect (src2, (mixer, 1))
f2c = gr.float_to_complex ()
fg.connect ( mixer, ( f2c, 0 ) );
usrp_interp = 400
u = usrp.sink_c (0, usrp_interp)
fg.connect(f2c, u)
return fg
def main():
fg = build_graph()
fg.start()
raw_input("Hit Enter to quit: ")
fg.stop()
if __name__ == '__main__':
main ()
Hi Josh - you almost have it. I've been wanting to work up an Am
wireless broadcaster
as an updated version of this:
http://oak.cats.ohiou.edu/~postr/bapix/KnightWB.htm
<http://oak.cats.ohiou.edu/%7Epostr/bapix/KnightWB.htm>
to play my own sound through old tube sets during the Christmas holidays.
A couple of points: The sampling_freq in the script is way too low for
the frequency you're asking for in src1. Looks like you're aiming for
1090 on the AM dial
so the sampling rate is going to need to be at least twice that per
Nyquist. Plus you have
to use a rate that, after usrp interpolation comes out 128Mhz. Also,
to create an AM signal
you mix an RF carrier with your AF plus a DC offset. Lastly - well,
in the maximally simple
script gr.float_to_complex() works since the usrp isn't doing any
frequency shifting.
You can directly make low AM broadcast band signals using a sampling
rate of 2Mhz
and usrp_interp of 64. The maximally simple script that works for me is:
#!/usr/bin/env python
from gnuradio import gr
from gnuradio import usrp
def build_graph ():
sampling_freq = 2e6
fg = gr.flow_graph ()
src0 = gr.sig_source_f ( sampling_freq, gr.GR_SIN_WAVE, 560e3,
4000, 0 )
src1 = gr.sig_source_f ( sampling_freq, gr.GR_SIN_WAVE, 440, .8, 1 )
mixer = gr.multiply_ff ()
dst = usrp.sink_c(0, 64)
f2c = gr.float_to_complex()
fg.connect ( src0, (mixer, 0) )
fg.connect ( src1, (mixer, 1) )
fg.connect ( mixer, f2c, dst )
return fg
if __name__ == '__main__':
fg = build_graph ()
fg.start ()
raw_input ('Press Enter to quit: ')
fg.stop ()
--------------------------------------------
Notice the 440hz audio tone has a DC offset=1. This actually works in
my lab with a
+10dbm amp and a tabletop am set. The amp has a 50ohm resistor load
and a cliplead
dangling near the transistor set tuned to 560Khz.
The next script uses usrp frequency translation so it MUST use a Hilbert
filter to get from float to complex - otherwise you get unwanted spurious
output signals. Using usrp frequency shifting you can work at a lower
sample
rate, but still create signals up to 1700Khz.
-----------------------------------------
#!/usr/bin/env python
#
# AM generation
#
# rf_loc
# |
# [audio]--[scale]--[offset]---[lpf2c]---(x)-----> AM out
# rf_mixc
#
#
from gnuradio import gr
from gnuradio import usrp
from gnuradio import audio_oss as audio
import sys
def build_graph ():
usrp_interp = 80
usrp_duc = 1050e3
sink = usrp.sink_c (0, usrp_interp)
print "requesting: ", usrp_duc, "hz"
sink.set_tx_freq(0, usrp_duc)
# use usrp.tx_freq(0) to get actual freq, and use that
# to set rf_LO
actual_freq = sink.tx_freq(0)
print "got ", actual_freq, "hz"
sink.set_nchannels(1)
sink.set_mux(0x98)
rf_LO = 40e3
af_sample_rate = 32000
rf_sample_rate = 1600000
fir_interp = rf_sample_rate / af_sample_rate
fg = gr.flow_graph ()
# transmit an audio file:
# audio_file = gr.file_source (gr.sizeof_short,
"/usr/src/tests/usrp_tests/work_11.sw",0)
# audio_conv = gr.short_to_float()
# ascale = gr.multiply_const_ff(.00003)
# src = gr.add_const_ff(1)
# fg.connect(audio_file, audio_conv, ascale, src)
# microphone/line-in source:
# src_mic = audio.source(af_sample_rate)
# ascale = gr.multiply_const_ff(100)
# src = gr.add_const_ff(1)
# fg.connect (src_mic, src)
# dial-tone test:
src1 = gr.sig_source_f (af_sample_rate,gr.GR_SIN_WAVE,440,.4,0)
src2 = gr.sig_source_f (af_sample_rate,gr.GR_SIN_WAVE,350,.4,0)
sum = gr.add_ff()
src = gr.add_const_ff(1)
fg.connect (src1, (sum, 0))
fg.connect (src2, (sum, 1))
fg.connect (sum, src)
lpf2_taps = gr.firdes.low_pass ( \
fir_interp, rf_sample_rate, 10e3, 10e3, gr.firdes.WIN_HAMMING)
lpf2c = gr.interp_fir_filter_fff (fir_interp, lpf2_taps)
rf_loc = gr.sig_source_f (rf_sample_rate,gr.GR_SIN_WAVE,rf_LO,2,0)
rf_mixc = gr.multiply_ff ()
scale = gr.multiply_const_ff(1e3)
hilbert = gr.hilbert_fc(97)
fg.connect (src, lpf2c)
fg.connect (lpf2c, (rf_mixc, 0))
fg.connect (rf_loc, (rf_mixc, 1))
fg.connect (rf_mixc, scale)
fg.connect (scale, hilbert)
fg.connect (hilbert, sink)
return fg
def main ():
fg = build_graph ()
fg.start ()
raw_input ('Press Enter to quit: ')
fg.stop ()
if __name__ == '__main__':
main ()
------------------------------------------------------------------------
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