diff options
author | Tom Rondeau <trondeau@vt.edu> | 2012-11-01 10:50:10 -0400 |
---|---|---|
committer | Tom Rondeau <trondeau@vt.edu> | 2012-11-01 10:50:10 -0400 |
commit | 1b83549181135e79ad00b9bae0d11ab27efa4148 (patch) | |
tree | 6b036dd9d26423528f49bad563c199a25a33f646 /gr-analog/python | |
parent | bbe599e4606b728c9610a17285b1e50f9e9d4dcb (diff) |
analog: removing PLL blocks. Also moved blks2impl files that used PLLs to gr-analog.
This probably impacts a lot of examples; once we've removed all files in gnuradio-core, we'll need to retest.
Diffstat (limited to 'gr-analog/python')
-rw-r--r-- | gr-analog/python/CMakeLists.txt | 10 | ||||
-rw-r--r-- | gr-analog/python/__init__.py | 11 | ||||
-rw-r--r-- | gr-analog/python/am_demod.py | 72 | ||||
-rw-r--r-- | gr-analog/python/fm_demod.py | 103 | ||||
-rw-r--r-- | gr-analog/python/fm_emph.py | 150 | ||||
-rw-r--r-- | gr-analog/python/nbfm_rx.py | 86 | ||||
-rw-r--r-- | gr-analog/python/nbfm_tx.py | 88 | ||||
-rw-r--r-- | gr-analog/python/standard_squelch.py | 77 | ||||
-rw-r--r-- | gr-analog/python/wfm_rcv.py | 68 | ||||
-rw-r--r-- | gr-analog/python/wfm_rcv_fmdet.py | 221 | ||||
-rw-r--r-- | gr-analog/python/wfm_rcv_pll.py | 188 | ||||
-rw-r--r-- | gr-analog/python/wfm_tx.py | 78 |
12 files changed, 1152 insertions, 0 deletions
diff --git a/gr-analog/python/CMakeLists.txt b/gr-analog/python/CMakeLists.txt index 08570eea33..f42ee20086 100644 --- a/gr-analog/python/CMakeLists.txt +++ b/gr-analog/python/CMakeLists.txt @@ -25,6 +25,16 @@ include(GrPython) GR_PYTHON_INSTALL( FILES __init__.py + am_demod.py + fm_demod.py + fm_emph.py + nbfm_rx.py + nbfm_tx.py + standard_squelch.py + wfm_rcv.py + wfm_rcv_fmdet.py + wfm_rcv_pll.py + wfm_tx.py DESTINATION ${GR_PYTHON_DIR}/gnuradio/analog COMPONENT "analog_python" ) diff --git a/gr-analog/python/__init__.py b/gr-analog/python/__init__.py index fc345028b9..a5ab5715d7 100644 --- a/gr-analog/python/__init__.py +++ b/gr-analog/python/__init__.py @@ -25,3 +25,14 @@ Blocks and utilities for analog modulation and demodulation. # The presence of this file turns this directory into a Python package from analog_swig import * + +from am_demod import * +from fm_demod import * +from fm_emph import * +from nbfm_rx import * +from nbfm_tx import * +from standard_squelch import * +from wfm_rcv import * +from wfm_rcv_fmdet import * +from wfm_rcv_pll import * +from wfm_tx import * diff --git a/gr-analog/python/am_demod.py b/gr-analog/python/am_demod.py new file mode 100644 index 0000000000..2730aa2622 --- /dev/null +++ b/gr-analog/python/am_demod.py @@ -0,0 +1,72 @@ +# +# Copyright 2006,2007,2012 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 filter + +class am_demod_cf(gr.hier_block2): + """ + Generalized AM demodulation block with audio filtering. + + This block demodulates a band-limited, complex down-converted AM + channel into the the original baseband signal, applying low pass + filtering to the audio output. It produces a float stream in the + range [-1.0, +1.0]. + + Args: + channel_rate: incoming sample rate of the AM baseband (integer) + audio_decim: input to output decimation rate (integer) + audio_pass: audio low pass filter passband frequency (float) + audio_stop: audio low pass filter stop frequency (float) + """ + def __init__(self, channel_rate, audio_decim, audio_pass, audio_stop): + gr.hier_block2.__init__(self, "am_demod_cf", + gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Input signature + + MAG = gr.complex_to_mag() + DCR = gr.add_const_ff(-1.0) + + audio_taps = filter.optfir.low_pass(0.5, # Filter gain + channel_rate, # Sample rate + audio_pass, # Audio passband + audio_stop, # Audio stopband + 0.1, # Passband ripple + 60) # Stopband attenuation + LPF = filter.fir_filter_fff(audio_decim, audio_taps) + + self.connect(self, MAG, DCR, LPF, self) + +class demod_10k0a3e_cf(am_demod_cf): + """ + AM demodulation block, 10 KHz channel. + + This block demodulates an AM channel conformant to 10K0A3E emission + standards, such as broadcast band AM transmissions. + + Args: + channel_rate: incoming sample rate of the AM baseband (integer) + audio_decim: input to output decimation rate (integer) + """ + def __init__(self, channel_rate, audio_decim): + am_demod_cf.__init__(self, channel_rate, audio_decim, + 5000, # Audio passband + 5500) # Audio stopband diff --git a/gr-analog/python/fm_demod.py b/gr-analog/python/fm_demod.py new file mode 100644 index 0000000000..c14ed1682e --- /dev/null +++ b/gr-analog/python/fm_demod.py @@ -0,0 +1,103 @@ +# +# Copyright 2006,2007,2012 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, analog, filter +from gnuradio.blks2impl.fm_emph import fm_deemph +from math import pi + +class fm_demod_cf(gr.hier_block2): + """ + Generalized FM demodulation block with deemphasis and audio + filtering. + + This block demodulates a band-limited, complex down-converted FM + channel into the the original baseband signal, optionally applying + deemphasis. Low pass filtering is done on the resultant signal. It + produces an output float strem in the range of [-1.0, +1.0]. + + Args: + channel_rate: incoming sample rate of the FM baseband (integer) + deviation: maximum FM deviation (default = 5000) (float) + audio_decim: input to output decimation rate (integer) + audio_pass: audio low pass filter passband frequency (float) + audio_stop: audio low pass filter stop frequency (float) + gain: gain applied to audio output (default = 1.0) (float) + tau: deemphasis time constant (default = 75e-6), specify 'None' to prevent deemphasis + """ + def __init__(self, channel_rate, audio_decim, deviation, + audio_pass, audio_stop, gain=1.0, tau=75e-6): + gr.hier_block2.__init__(self, "fm_demod_cf", + gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + k = channel_rate/(2*pi*deviation) + QUAD = analog.quadrature_demod_cf(k) + + audio_taps = filter.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 = filter.fir_filter_fff(audio_decim, audio_taps) + + if tau is not None: + DEEMPH = fm_deemph(channel_rate, tau) + self.connect(self, QUAD, DEEMPH, LPF, self) + else: + self.connect(self, QUAD, LPF, self) + +class demod_20k0f3e_cf(fm_demod_cf): + """ + NBFM demodulation block, 20 KHz channels + + This block demodulates a complex, downconverted, narrowband FM + channel conforming to 20K0F3E emission standards, outputting + floats in the range [-1.0, +1.0]. + + Args: + sample_rate: incoming sample rate of the FM baseband (integer) + audio_decim: input to output decimation rate (integer) + """ + def __init__(self, channel_rate, audio_decim): + fm_demod_cf.__init__(self, channel_rate, audio_decim, + 5000, # Deviation + 3000, # Audio passband frequency + 4500) # Audio stopband frequency + +class demod_200kf3e_cf(fm_demod_cf): + """ + WFM demodulation block, mono. + + This block demodulates a complex, downconverted, wideband FM + channel conforming to 200KF3E emission standards, outputting + floats in the range [-1.0, +1.0]. + + Args: + sample_rate: incoming sample rate of the FM baseband (integer) + audio_decim: input to output decimation rate (integer) + """ + def __init__(self, channel_rate, audio_decim): + fm_demod_cf.__init__(self, channel_rate, audio_decim, + 75000, # Deviation + 15000, # Audio passband + 16000, # Audio stopband + 20.0) # Audio gain diff --git a/gr-analog/python/fm_emph.py b/gr-analog/python/fm_emph.py new file mode 100644 index 0000000000..2821f6e3cd --- /dev/null +++ b/gr-analog/python/fm_emph.py @@ -0,0 +1,150 @@ +# +# Copyright 2005,2007,2012 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, filter +import math + +# +# 1 +# H(s) = ------- +# 1 + s +# +# tau is the RC time constant. +# critical frequency: w_p = 1/tau +# +# We prewarp and use the bilinear z-transform to get our IIR coefficients. +# See "Digital Signal Processing: A Practical Approach" by Ifeachor and Jervis +# + +class fm_deemph(gr.hier_block2): + """ + FM Deemphasis IIR filter. + """ + + + def __init__(self, fs, tau=75e-6): + """ + + Args: + fs: sampling frequency in Hz (float) + tau: Time constant in seconds (75us in US, 50us in EUR) (float) + """ + gr.hier_block2.__init__(self, "fm_deemph", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + w_p = 1/tau + w_pp = math.tan(w_p / (fs * 2)) # prewarped analog freq + + a1 = (w_pp - 1)/(w_pp + 1) + b0 = w_pp/(1 + w_pp) + b1 = b0 + + btaps = [b0, b1] + ataps = [1, a1] + + if 0: + print "btaps =", btaps + print "ataps =", ataps + global plot1 + plot1 = gru.gnuplot_freqz(gru.freqz(btaps, ataps), fs, True) + + deemph = filter.iir_filter_ffd(btaps, ataps) + self.connect(self, deemph, self) + +# +# 1 + s*t1 +# H(s) = ---------- +# 1 + s*t2 +# +# I think this is the right transfer function. +# +# +# This fine ASCII rendition is based on Figure 5-15 +# in "Digital and Analog Communication Systems", Leon W. Couch II +# +# +# R1 +# +-----||------+ +# | | +# o------+ +-----+--------o +# | C1 | | +# +----/\/\/\/--+ \ +# / +# \ R2 +# / +# \ +# | +# o--------------------------+--------o +# +# f1 = 1/(2*pi*t1) = 1/(2*pi*R1*C) +# +# 1 R1 + R2 +# f2 = ------- = ------------ +# 2*pi*t2 2*pi*R1*R2*C +# +# t1 is 75us in US, 50us in EUR +# f2 should be higher than our audio bandwidth. +# +# +# The Bode plot looks like this: +# +# +# /---------------- +# / +# / <-- slope = 20dB/decade +# / +# -------------/ +# f1 f2 +# +# We prewarp and use the bilinear z-transform to get our IIR coefficients. +# See "Digital Signal Processing: A Practical Approach" by Ifeachor and Jervis +# + +class fm_preemph(gr.hier_block2): + """ + FM Preemphasis IIR filter. + """ + def __init__(self, fs, tau=75e-6): + """ + + Args: + fs: sampling frequency in Hz (float) + tau: Time constant in seconds (75us in US, 50us in EUR) (float) + """ + + gr.hier_block2.__init__(self, "fm_deemph", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + # FIXME make this compute the right answer + + btaps = [1] + ataps = [1] + + if 0: + print "btaps =", btaps + print "ataps =", ataps + global plot2 + plot2 = gru.gnuplot_freqz(gru.freqz(btaps, ataps), fs, True) + + preemph = filter.iir_filter_ffd(btaps, ataps) + self.connect(self, preemph, self) diff --git a/gr-analog/python/nbfm_rx.py b/gr-analog/python/nbfm_rx.py new file mode 100644 index 0000000000..1d307328b1 --- /dev/null +++ b/gr-analog/python/nbfm_rx.py @@ -0,0 +1,86 @@ +# +# Copyright 2005,2012 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. +# + +import math +from gnuradio import gr +from gnuradio import filter +from gnuradio import analog +import fm_deemph + +class nbfm_rx(gr.hier_block2): + def __init__(self, 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]. + + Args: + audio_rate: sample rate of audio stream, >= 16k (integer) + quad_rate: sample rate of output stream (integer) + tau: preemphasis time constant (default 75e-6) (float) + max_dev: maximum deviation in Hz (default 5e3) (float) + + quad_rate must be an integer multiple of audio_rate. + + Exported sub-blocks (attributes): + squelch + quad_demod + deemph + audio_filter + """ + + gr.hier_block2.__init__(self, "nbfm_rx", + gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + # 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 = analog.quadrature_demod_cf(k) + + # FM Deemphasis IIR filter + self.deemph = fm_deemph(quad_rate, tau=tau) + + # compute FIR taps for audio filter + audio_decim = quad_rate // audio_rate + audio_taps = filter.firdes.low_pass(1.0, # gain + quad_rate, # sampling rate + 2.7e3, # Audio LPF cutoff + 0.5e3, # 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 = filter.fir_filter_fff(audio_decim, audio_taps) + + self.connect(self, self.quad_demod, self.deemph, self.audio_filter, self) diff --git a/gr-analog/python/nbfm_tx.py b/gr-analog/python/nbfm_tx.py new file mode 100644 index 0000000000..a108aac49e --- /dev/null +++ b/gr-analog/python/nbfm_tx.py @@ -0,0 +1,88 @@ +# +# Copyright 2005,2012 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. +# + +import math +from gnuradio import gr, filter +from gnuradio import analog +from fm_emph import fm_preemph + +class nbfm_tx(gr.hier_block2): + def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=5e3): + """ + Narrow Band FM Transmitter. + + Takes a single float input stream of audio samples in the range [-1,+1] + and produces a single FM modulated complex baseband output. + + Args: + audio_rate: sample rate of audio stream, >= 16k (integer) + quad_rate: sample rate of output stream (integer) + tau: preemphasis time constant (default 75e-6) (float) + max_dev: maximum deviation in Hz (default 5e3) (float) + + quad_rate must be an integer multiple of audio_rate. + """ + + gr.hier_block2.__init__(self, "nbfm_tx", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_gr_complex)) # Output signature + + # 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" + + + do_interp = audio_rate != quad_rate + + if do_interp: + interp_factor = quad_rate / audio_rate + interp_taps = filter.optfir.low_pass(interp_factor, # gain + quad_rate, # Fs + 4500, # passband cutoff + 7000, # stopband cutoff + 0.1, # passband ripple dB + 40) # stopband atten dB + + #print "len(interp_taps) =", len(interp_taps) + self.interpolator = filter.interp_fir_filter_fff (interp_factor, interp_taps) + + self.preemph = fm_preemph(quad_rate, tau=tau) + + k = 2 * math.pi * max_dev / quad_rate + self.modulator = analog.frequency_modulator_fc(k) + + if do_interp: + self.connect(self, self.interpolator, self.preemph, self.modulator, self) + else: + self.connect(self, self.preemph, self.modulator, self) + + +class ctcss_gen_f(gr.hier_block2): + def __init__(self, sample_rate, tone_freq): + gr.hier_block2.__init__(self, "ctcss_gen_f", + gr.io_signature(0, 0, 0), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + self.plgen = analog.sig_source_f(sample_rate, gr.GR_SIN_WAVE, tone_freq, 0.1, 0.0) + self.connect(self.plgen, self) diff --git a/gr-analog/python/standard_squelch.py b/gr-analog/python/standard_squelch.py new file mode 100644 index 0000000000..986397a48e --- /dev/null +++ b/gr-analog/python/standard_squelch.py @@ -0,0 +1,77 @@ +# +# Copyright 2005,2007,2012 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. +# + +import math +from gnuradio import gr +from gnuradio import filter + +class standard_squelch(gr.hier_block2): + def __init__(self, audio_rate): + gr.hier_block2.__init__(self, "standard_squelch", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + self.input_node = gr.add_const_ff(0) # FIXME kludge + + self.low_iir = filter.iir_filter_ffd((0.0193,0,-0.0193),(1,1.9524,-0.9615)) + self.low_square = gr.multiply_ff() + self.low_smooth = filter.single_pole_iir_filter_ff(1/(0.01*audio_rate)) # 100ms time constant + + self.hi_iir = filter.iir_filter_ffd((0.0193,0,-0.0193),(1,1.3597,-0.9615)) + self.hi_square = gr.multiply_ff() + self.hi_smooth = filter.single_pole_iir_filter_ff(1/(0.01*audio_rate)) + + self.sub = gr.sub_ff(); + self.add = gr.add_ff(); + self.gate = gr.threshold_ff(0.3,0.43,0) + self.squelch_lpf = filter.single_pole_iir_filter_ff(1/(0.01*audio_rate)) + + self.div = gr.divide_ff() + self.squelch_mult = gr.multiply_ff() + + self.connect(self, self.input_node) + self.connect(self.input_node, (self.squelch_mult, 0)) + + self.connect(self.input_node,self.low_iir) + self.connect(self.low_iir,(self.low_square,0)) + self.connect(self.low_iir,(self.low_square,1)) + self.connect(self.low_square,self.low_smooth,(self.sub,0)) + self.connect(self.low_smooth, (self.add,0)) + + self.connect(self.input_node,self.hi_iir) + self.connect(self.hi_iir,(self.hi_square,0)) + self.connect(self.hi_iir,(self.hi_square,1)) + self.connect(self.hi_square,self.hi_smooth,(self.sub,1)) + self.connect(self.hi_smooth, (self.add,1)) + + self.connect(self.sub, (self.div, 0)) + self.connect(self.add, (self.div, 1)) + self.connect(self.div, self.gate, self.squelch_lpf, (self.squelch_mult,1)) + self.connect(self.squelch_mult, self) + + def set_threshold(self, threshold): + self.gate.set_hi(threshold) + + def threshold(self): + return self.gate.hi() + + def squelch_range(self): + return (0.0, 1.0, 1.0/100) diff --git a/gr-analog/python/wfm_rcv.py b/gr-analog/python/wfm_rcv.py new file mode 100644 index 0000000000..addbb1d258 --- /dev/null +++ b/gr-analog/python/wfm_rcv.py @@ -0,0 +1,68 @@ +# +# Copyright 2005,2007,2012 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, analog, filter +from gnuradio.blks2impl.fm_emph import fm_deemph +import math + +class wfm_rcv(gr.hier_block2): + def __init__ (self, 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). + + Args: + quad_rate: input sample rate of complex baseband input. (float) + audio_decimation: how much to decimate quad_rate to get to audio. (integer) + """ + gr.hier_block2.__init__(self, "wfm_rcv", + gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature + gr.io_signature(1, 1, gr.sizeof_float)) # Output signature + + 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 = analog.quadrature_demod_cf(fm_demod_gain) + + # input: float; output: float + self.deemph = 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 + quad_rate, # sampling rate + audio_rate/2 - width_of_transition_band, + width_of_transition_band, + filter.firdes.WIN_HAMMING) + # input: float; output: float + self.audio_filter = filter.fir_filter_fff(audio_decimation, audio_coeffs) + + self.connect (self, self.fm_demod, self.audio_filter, self.deemph, self) diff --git a/gr-analog/python/wfm_rcv_fmdet.py b/gr-analog/python/wfm_rcv_fmdet.py new file mode 100644 index 0000000000..68c7a3a1d7 --- /dev/null +++ b/gr-analog/python/wfm_rcv_fmdet.py @@ -0,0 +1,221 @@ +# +# Copyright 2005,2006,2012 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, filter, analog +from gnuradio.blks2impl.fm_emph import fm_deemph +import math + +class wfm_rcv_fmdet(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_fmdet", + gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature + gr.io_signature(2, 2, gr.sizeof_float)) # Output signature + lowfreq = -125e3/demod_rate + highfreq = 125e3/demod_rate + 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 + + self.fm_demod = analog.fmdet_cf(demod_rate, lowfreq, highfreq, 0.05) + + # 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 = gr.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 = gr.multiply_cc(); + self.rds_signal_generator = gr.multiply_cc(); + self_rds_signal_processor = gr.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 = 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 = \ + 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 + # the resulting signal from this multiplier is the carrier + # with correct phase but at -38000 Hz. + self.connect(self.stereo_carrier_pll_recovery, (self.stereo_carrier_generator,1)) + + # 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 + # the result is BASEBANDED DSBSC with phase zero! + self.connect(self.fm_demod,self.stereo_dsbsc_filter, (self.stereo_basebander,1)) + + # 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) diff --git a/gr-analog/python/wfm_rcv_pll.py b/gr-analog/python/wfm_rcv_pll.py new file mode 100644 index 0000000000..953cd538f1 --- /dev/null +++ b/gr-analog/python/wfm_rcv_pll.py @@ -0,0 +1,188 @@ +# +# Copyright 2005,2006,2012 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, filter, analog +from gnuradio.blks2impl.fm_emph import fm_deemph +import math + +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 = gr.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 = gr.multiply_cc(); + self.rds_signal_generator = gr.multiply_cc(); + self_rds_signal_processor = gr.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 = 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 = \ + 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) diff --git a/gr-analog/python/wfm_tx.py b/gr-analog/python/wfm_tx.py new file mode 100644 index 0000000000..4c074b5317 --- /dev/null +++ b/gr-analog/python/wfm_tx.py @@ -0,0 +1,78 @@ +# +# Copyright 2005,2007,2012 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. +# + +import math +from gnuradio import gr +from gnuradio import filter +from gnuradio import analog +from fm_emph import fm_preemph + +class wfm_tx(gr.hier_block2): + def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=75e3): + """ + Wide Band FM Transmitter. + + Takes a single float input stream of audio samples in the range [-1,+1] + and produces a single FM modulated complex baseband output. + + Args: + audio_rate: sample rate of audio stream, >= 16k (integer) + quad_rate: sample rate of output stream (integer) + tau: preemphasis time constant (default 75e-6) (float) + max_dev: maximum deviation in Hz (default 75e3) (float) + + quad_rate must be an integer multiple of audio_rate. + """ + gr.hier_block2.__init__(self, "wfm_tx", + gr.io_signature(1, 1, gr.sizeof_float), # Input signature + gr.io_signature(1, 1, gr.sizeof_gr_complex)) # Output signature + + # 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" + + + do_interp = audio_rate != quad_rate + + if do_interp: + interp_factor = quad_rate / audio_rate + interp_taps = filter.optfir.low_pass(interp_factor, # gain + quad_rate, # Fs + 16000, # passband cutoff + 18000, # stopband cutoff + 0.1, # passband ripple dB + 40) # stopband atten dB + + print "len(interp_taps) =", len(interp_taps) + self.interpolator = filter.interp_fir_filter_fff (interp_factor, interp_taps) + + self.preemph = fm_preemph(quad_rate, tau=tau) + + k = 2 * math.pi * max_dev / quad_rate + self.modulator = analog.frequency_modulator_fc (k) + + if do_interp: + self.connect(self, self.interpolator, self.preemph, self.modulator, self) + else: + self.connect(self, self.preemph, self.modulator, self) |