analog: pep8 formatting

Signed-off-by: Josh Morman <jmorman@gnuradio.org>
author: Josh Morman <jmorman@gnuradio.org> 2021-11-24 12:01:00 -0500
committer: mormj <34754695+mormj@users.noreply.github.com> 2021-11-24 14:41:53 -0500
commit: 2ede969ec5f3f6d361814403b7375db5f15ec14d (patch)
tree: d963e92732842ee3491da1282dfa09400342c96d /gr-analog/python
parent: c6f291d57d913708a2517de3c6902425c2c575d3 (diff)
11 files changed, 260 insertions, 194 deletions
diff --git a/gr-analog/python/analog/am_demod.py b/gr-analog/python/analog/am_demod.py
index a3c2a8e9de..34b7ddaf74 100644
--- a/gr-analog/python/analog/am_demod.py
+++ b/gr-analog/python/analog/am_demod.py
@@ -11,6 +11,7 @@ from gnuradio import gr
 from gnuradio import blocks
 from gnuradio import filter
 
+
 class am_demod_cf(gr.hier_block2):
     """
     Generalized AM demodulation block with audio filtering.
@@ -26,16 +27,18 @@ class am_demod_cf(gr.hier_block2):
         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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(1, 1, gr.sizeof_float))      # Input signature
 
         MAG = blocks.complex_to_mag()
         DCR = blocks.add_const_ff(-1.0)
 
         audio_taps = filter.optfir.low_pass(0.5,          # Filter gain
-                                            channel_rate, # Sample rate
+                                            channel_rate,  # Sample rate
                                             audio_pass,   # Audio passband
                                             audio_stop,   # Audio stopband
                                             0.1,          # Passband ripple
@@ -44,6 +47,7 @@ class am_demod_cf(gr.hier_block2):
 
         self.connect(self, MAG, DCR, LPF, self)
 
+
 class demod_10k0a3e_cf(am_demod_cf):
     """
     AM demodulation block, 10 KHz channel.
@@ -55,7 +59,8 @@ class demod_10k0a3e_cf(am_demod_cf):
         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
+                             5000,  # Audio passband
+                             5500)  # Audio stopband
diff --git a/gr-analog/python/analog/fm_demod.py b/gr-analog/python/analog/fm_demod.py
index 96fed682ff..e829871df8 100644
--- a/gr-analog/python/analog/fm_demod.py
+++ b/gr-analog/python/analog/fm_demod.py
@@ -34,13 +34,15 @@ class fm_demod_cf(gr.hier_block2):
         gain: gain applied to audio output (default = 1.0) (float)
         tau: deemphasis time constant (default = 75e-6), specify tau=0.0 to prevent deemphasis (float)
     """
+
     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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(1, 1, gr.sizeof_float))       # Output signature
 
-        k = channel_rate / (2*pi*deviation)
+        k = channel_rate / (2 * pi * deviation)
         QUAD = analog.quadrature_demod_cf(k)
 
         audio_taps = filter.optfir.low_pass(
@@ -72,6 +74,7 @@ class demod_20k0f3e_cf(fm_demod_cf):
         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
@@ -91,6 +94,7 @@ class demod_200kf3e_cf(fm_demod_cf):
         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
diff --git a/gr-analog/python/analog/fm_emph.py b/gr-analog/python/analog/fm_emph.py
index 9c1107bb14..a581bdf2ac 100644
--- a/gr-analog/python/analog/fm_emph.py
+++ b/gr-analog/python/analog/fm_emph.py
@@ -16,22 +16,22 @@ import cmath
 class fm_deemph(gr.hier_block2):
     r"""
     FM Deemphasis IIR filter
-    
+
     Args:
         fs: sampling frequency in Hz (float)
         tau: Time constant in seconds (75us in US, 50us in EUR) (float)
-        
+
   An analog deemphasis filter:
- 
+
                R
   o------/\/\/\/---+----o
                    |
                   = C
                    |
                   ---
- 
+
   Has this transfer function:
- 
+
                1             1
               ----          ---
                RC          tau
@@ -39,39 +39,39 @@ class fm_deemph(gr.hier_block2):
                  1             1
             s + ----      s + ---
                  RC           tau
- 
+
   And has its -3 dB response, due to the pole, at
- 
+
   |H(j w_c)|^2 = 1/2  =>  s = j w_c = j (1/(RC))
- 
+
   Historically, this corner frequency of analog audio deemphasis filters
   been specified by the RC time constant used, called tau.
   So w_c = 1/tau.
- 
+
   FWIW, for standard tau values, some standard analog components would be:
   tau = 75 us = (50K)(1.5 nF) = (50 ohms)(1.5 uF)
   tau = 50 us = (50K)(1.0 nF) = (50 ohms)(1.0 uF)
- 
+
   In specifying tau for this digital deemphasis filter, tau specifies
   the *digital* corner frequency, w_c, desired.
- 
+
   The digital deemphasis filter design below, uses the
   "bilinear transformation" method of designing digital filters:
- 
+
   1. Convert digital specifications into the analog domain, by prewarping
      digital frequency specifications into analog frequencies.
- 
+
      w_a = (2/T)tan(wT/2)
- 
+
   2. Use an analog filter design technique to design the filter.
- 
+
   3. Use the bilinear transformation to convert the analog filter design to a
      digital filter design.
- 
+
      H(z) = H(s)|
                      s = (2/T)(1-z^-1)/(1+z^-1)
- 
- 
+
+
          w_ca         1          1 - (-1) z^-1
   H(z) = ---- * ----------- * -----------------------
          2 fs        -w_ca             -w_ca
@@ -81,20 +81,21 @@ class fm_deemph(gr.hier_block2):
                                        -w_ca
                                    1 - -----
                                         2 fs
- 
+
   We use this design technique, because it is an easy way to obtain a filter
   design with the -6 dB/octave roll-off required of the deemphasis filter.
- 
+
   Jackson, Leland B., _Digital_Filters_and_Signal_Processing_Second_Edition_,
     Kluwer Academic Publishers, 1989, pp 201-212
- 
+
   Orfanidis, Sophocles J., _Introduction_to_Signal_Processing_, Prentice Hall,
     1996, pp 573-583
     """
 
     def __init__(self, fs, tau=75e-6):
         gr.hier_block2.__init__(self, "fm_deemph",
-                                gr.io_signature(1, 1, gr.sizeof_float),  # Input signature
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_float),
                                 gr.io_signature(1, 1, gr.sizeof_float))  # Output signature
 
         # Digital corner frequency
@@ -111,8 +112,8 @@ class fm_deemph(gr.hier_block2):
         p1 = (1.0 + k) / (1.0 - k)
         b0 = -k / (1.0 - k)
 
-        btaps = [ b0 * 1.0, b0 * -z1 ]
-        ataps = [      1.0,      -p1 ]
+        btaps = [b0 * 1.0, b0 * -z1]
+        ataps = [1.0, -p1]
 
         # Since H(s = 0) = 1.0, then H(z = 1) = 1.0 and has 0 dB gain at DC
 
@@ -120,18 +121,17 @@ class fm_deemph(gr.hier_block2):
         self.connect(self, deemph, self)
 
 
-
 class fm_preemph(gr.hier_block2):
     r"""
     FM Preemphasis IIR filter.
-    
+
     Args:
         fs: sampling frequency in Hz (float)
         tau: Time constant in seconds (75us in US, 50us in EUR) (float)
-        fh: High frequency at which to flatten out (< 0 means default of 0.925*fs/2.0) (float)    
-    
+        fh: High frequency at which to flatten out (< 0 means default of 0.925*fs/2.0) (float)
+
   An analog preemphasis filter, that flattens out again at the high end:
- 
+
                C
          +-----||------+
          |             |
@@ -144,12 +144,12 @@ class fm_preemph(gr.hier_block2):
                              \
                              |
   o--------------------------+--------o
- 
+
   (This fine ASCII rendition is based on Figure 5-15
    in "Digital and Analog Communication Systems", Leon W. Couch II)
- 
+
   Has this transfer function:
- 
+
                    1
               s + ---
                   R1C
@@ -157,58 +157,58 @@ class fm_preemph(gr.hier_block2):
                1       R1
           s + --- (1 + --)
               R1C      R2
- 
- 
+
+
   It has a corner due to the numerator, where the rise starts, at
- 
+
   |Hn(j w_cl)|^2 = 2*|Hn(0)|^2  =>  s = j w_cl = j (1/(R1C))
- 
+
   It has a corner due to the denominator, where it levels off again, at
- 
+
   |Hn(j w_ch)|^2 = 1/2*|Hd(0)|^2  =>  s = j w_ch = j (1/(R1C) * (1 + R1/R2))
- 
+
   Historically, the corner frequency of analog audio preemphasis filters
   been specified by the R1C time constant used, called tau.
- 
+
   So
   w_cl = 1/tau  =         1/R1C; f_cl = 1/(2*pi*tau)  =         1/(2*pi*R1*C)
   w_ch = 1/tau2 = (1+R1/R2)/R1C; f_ch = 1/(2*pi*tau2) = (1+R1/R2)/(2*pi*R1*C)
- 
+
   and note f_ch = f_cl * (1 + R1/R2).
- 
+
   For broadcast FM audio, tau is 75us in the United States and 50us in Europe.
   f_ch should be higher than our digital audio bandwidth.
- 
+
   The Bode plot looks like this:
- 
- 
+
+
                      /----------------
                     /
                    /  <-- slope = 20dB/decade
                   /
     -------------/
                f_cl  f_ch
- 
+
   In specifying tau for this digital preemphasis filter, tau specifies
   the *digital* corner frequency, w_cl, desired.
- 
+
   The digital preemphasis filter design below, uses the
   "bilinear transformation" method of designing digital filters:
- 
+
   1. Convert digital specifications into the analog domain, by prewarping
      digital frequency specifications into analog frequencies.
- 
+
      w_a = (2/T)tan(wT/2)
- 
+
   2. Use an analog filter design technique to design the filter.
- 
+
   3. Use the bilinear transformation to convert the analog filter design to a
      digital filter design.
- 
+
      H(z) = H(s)|
                  s = (2/T)(1-z^-1)/(1+z^-1)
- 
- 
+
+
                                   -w_cla
                               1 + ------
                                    2 fs
@@ -224,19 +224,21 @@ class fm_preemph(gr.hier_block2):
                                   -w_cha
                               1 - ------
                                    2 fs
- 
+
   We use this design technique, because it is an easy way to obtain a filter
   design with the 6 dB/octave rise required of the premphasis filter.
- 
+
   Jackson, Leland B., _Digital_Filters_and_Signal_Processing_Second_Edition_,
     Kluwer Academic Publishers, 1989, pp 201-212
- 
+
   Orfanidis, Sophocles J., _Introduction_to_Signal_Processing_, Prentice Hall,
     1996, pp 573-583
     """
+
     def __init__(self, fs, tau=75e-6, fh=-1.0):
         gr.hier_block2.__init__(self, "fm_preemph",
-                                gr.io_signature(1, 1, gr.sizeof_float),  # Input signature
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_float),
                                 gr.io_signature(1, 1, gr.sizeof_float))  # Output signature
 
         # Set fh to something sensible, if needed.
@@ -244,7 +246,7 @@ class fm_preemph(gr.hier_block2):
         # at z = -1.0 or z = 1.0 respectively.  That makes the filter unstable
         # and useless.
         if fh <= 0.0 or fh >= fs / 2.0:
-            fh = 0.925 * fs/2.0
+            fh = 0.925 * fs / 2.0
 
         # Digital corner frequencies
         w_cl = 1.0 / tau
@@ -268,11 +270,11 @@ class fm_preemph(gr.hier_block2):
         # That isn't what users are going to expect, so adjust with a
         # gain, g, so that H(z = 1) = 1.0 for 0 dB gain at DC.
         w_0dB = 2.0 * math.pi * 0.0
-        g =        abs(1.0 - p1 * cmath.rect(1.0, -w_0dB)) \
+        g = abs(1.0 - p1 * cmath.rect(1.0, -w_0dB)) \
             / (b0 * abs(1.0 - z1 * cmath.rect(1.0, -w_0dB)))
 
-        btaps = [ g * b0 * 1.0, g * b0 * -z1 ]
-        ataps = [          1.0,          -p1 ]
+        btaps = [g * b0 * 1.0, g * b0 * -z1]
+        ataps = [1.0, -p1]
 
         preemph = filter.iir_filter_ffd(btaps, ataps, False)
         self.connect(self, preemph, self)
diff --git a/gr-analog/python/analog/nbfm_rx.py b/gr-analog/python/analog/nbfm_rx.py
index 72b2dcc936..232c8ab725 100644
--- a/gr-analog/python/analog/nbfm_rx.py
+++ b/gr-analog/python/analog/nbfm_rx.py
@@ -38,9 +38,11 @@ class nbfm_rx(gr.hier_block2):
       deemph
       audio_filter
     """
+
     def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=5e3):
         gr.hier_block2.__init__(self, "nbfm_rx",
-                                gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(1, 1, gr.sizeof_float))      # Output signature
 
         # FIXME audio_rate and quad_rate ought to be exact rationals
@@ -48,13 +50,14 @@ class nbfm_rx(gr.hier_block2):
         self._quad_rate = quad_rate = int(quad_rate)
 
         if quad_rate % audio_rate != 0:
-            raise ValueError("quad_rate is not an integer multiple of audio_rate")
+            raise ValueError(
+                "quad_rate is not an integer multiple of audio_rate")
 
         squelch_threshold = 20        # dB
         #self.squelch = analog.simple_squelch_cc(squelch_threshold, 0.001)
 
         # FM Demodulator  input: complex; output: float
-        k = quad_rate / (2*math.pi*max_dev)
+        k = quad_rate / (2 * math.pi * max_dev)
         self.quad_demod = analog.quadrature_demod_cf(k)
 
         # FM Deemphasis IIR filter
@@ -74,8 +77,9 @@ class nbfm_rx(gr.hier_block2):
         # 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)
+        self.connect(self, self.quad_demod, self.deemph,
+                     self.audio_filter, self)
 
     def set_max_deviation(self, max_dev):
-        k = self._quad_rate / (2*math.pi*max_dev)
+        k = self._quad_rate / (2 * math.pi * max_dev)
         self.quad_demod.set_gain(k)
diff --git a/gr-analog/python/analog/nbfm_tx.py b/gr-analog/python/analog/nbfm_tx.py
index df7c6c6a58..653fb2c5dc 100644
--- a/gr-analog/python/analog/nbfm_tx.py
+++ b/gr-analog/python/analog/nbfm_tx.py
@@ -32,23 +32,26 @@ class nbfm_tx(gr.hier_block2):
 
     quad_rate must be an integer multiple of audio_rate.
     """
+
     def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=5e3, fh=-1.0):
         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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_float),
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex))  # Output signature
 
         # FIXME audio_rate and quad_rate ought to be exact rationals
         self._audio_rate = audio_rate = int(audio_rate)
         self._quad_rate = quad_rate = int(quad_rate)
 
         if quad_rate % audio_rate != 0:
-            raise ValueError("quad_rate is not an integer multiple of audio_rate")
-
+            raise ValueError(
+                "quad_rate is not an integer multiple of audio_rate")
 
         do_interp = audio_rate != quad_rate
 
         if do_interp:
-            interp_factor = int(quad_rate / audio_rate)           # force integer
+            # force integer
+            interp_factor = int(quad_rate / audio_rate)
             interp_taps = filter.optfir.low_pass(interp_factor,   # gain
                                                  quad_rate,       # Fs
                                                  4500,            # passband cutoff
@@ -57,7 +60,8 @@ class nbfm_tx(gr.hier_block2):
                                                  40)              # stopband atten dB
 
             #print("len(interp_taps) =", len(interp_taps))
-            self.interpolator = filter.interp_fir_filter_fff (interp_factor, interp_taps)
+            self.interpolator = filter.interp_fir_filter_fff(
+                interp_factor, interp_taps)
 
         self.preemph = fm_preemph(quad_rate, tau=tau, fh=fh)
 
@@ -65,7 +69,8 @@ class nbfm_tx(gr.hier_block2):
         self.modulator = analog.frequency_modulator_fc(k)
 
         if do_interp:
-            self.connect(self, self.interpolator, self.preemph, self.modulator, self)
+            self.connect(self, self.interpolator,
+                         self.preemph, self.modulator, self)
         else:
             self.connect(self, self.preemph, self.modulator, self)
 
@@ -74,12 +79,12 @@ class nbfm_tx(gr.hier_block2):
         self.modulator.set_sensitivity(k)
 
 
-
 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
+                                # Input signature
+                                gr.io_signature(0, 0, 0),
+                                gr.io_signature(1, 1, gr.sizeof_float))  # Output signature
 
         self.plgen = analog.sig_source_f(sample_rate, analog.GR_SIN_WAVE,
                                          tone_freq, 0.1, 0.0)
diff --git a/gr-analog/python/analog/qa_agc.py b/gr-analog/python/analog/qa_agc.py
index 9368dd5d2a..b6d1f8957b 100644
--- a/gr-analog/python/analog/qa_agc.py
+++ b/gr-analog/python/analog/qa_agc.py
@@ -369,7 +369,7 @@ class test_agc(gr_unittest.TestCase):
         tb = self.tb
 
         expected_result = \
-            ((100+0j),
+            ((100 + 0j),
              (0.8090173602104187 + 0.5877856016159058j),
              (0.3090175688266754 + 0.9510582685470581j),
              (-0.309017539024353 + 0.9510582089424133j),
diff --git a/gr-analog/python/analog/standard_squelch.py b/gr-analog/python/analog/standard_squelch.py
index 49c4b589b7..e70efce700 100644
--- a/gr-analog/python/analog/standard_squelch.py
+++ b/gr-analog/python/analog/standard_squelch.py
@@ -12,26 +12,33 @@ from gnuradio import gr
 from gnuradio import blocks
 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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_float),
+                                gr.io_signature(1, 1, gr.sizeof_float))  # Output signature
 
         self.input_node = blocks.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_iir = filter.iir_filter_ffd(
+            (0.0193, 0, -0.0193), (1, 1.9524, -0.9615))
         self.low_square = blocks.multiply_ff()
-        self.low_smooth = filter.single_pole_iir_filter_ff(1 / (0.01*audio_rate))   # 100ms time constant
+        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_iir = filter.iir_filter_ffd(
+            (0.0193, 0, -0.0193), (1, 1.3597, -0.9615))
         self.hi_square = blocks.multiply_ff()
-        self.hi_smooth = filter.single_pole_iir_filter_ff(1 / (0.01*audio_rate))
+        self.hi_smooth = filter.single_pole_iir_filter_ff(
+            1 / (0.01 * audio_rate))
 
-        self.sub = blocks.sub_ff();
-        self.add = blocks.add_ff();
-        self.gate = blocks.threshold_ff(0.3,0.43,0)
-        self.squelch_lpf = filter.single_pole_iir_filter_ff(1 / (0.01*audio_rate))
+        self.sub = blocks.sub_ff()
+        self.add = blocks.add_ff()
+        self.gate = blocks.threshold_ff(0.3, 0.43, 0)
+        self.squelch_lpf = filter.single_pole_iir_filter_ff(
+            1 / (0.01 * audio_rate))
 
         self.div = blocks.divide_ff()
         self.squelch_mult = blocks.multiply_ff()
@@ -39,21 +46,22 @@ class standard_squelch(gr.hier_block2):
         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.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.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.div, self.gate, self.squelch_lpf,
+                     (self.squelch_mult, 1))
         self.connect(self.squelch_mult, self)
 
     def set_threshold(self, threshold):
diff --git a/gr-analog/python/analog/wfm_rcv.py b/gr-analog/python/analog/wfm_rcv.py
index 34ba9787db..d6e81ea7cd 100644
--- a/gr-analog/python/analog/wfm_rcv.py
+++ b/gr-analog/python/analog/wfm_rcv.py
@@ -17,7 +17,7 @@ from .fm_emph import fm_deemph
 
 
 class wfm_rcv(gr.hier_block2):
-    def __init__ (self, quad_rate, audio_decimation):
+    def __init__(self, quad_rate, audio_decimation):
         """
         Hierarchical block for demodulating a broadcast FM signal.
 
@@ -29,7 +29,8 @@ class wfm_rcv(gr.hier_block2):
             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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(1, 1, gr.sizeof_float))      # Output signature
         if audio_decimation != int(audio_decimation):
             raise ValueError("audio_decimation needs to be an integer")
@@ -38,10 +39,9 @@ class wfm_rcv(gr.hier_block2):
         volume = 20.
 
         max_dev = 75e3
-        fm_demod_gain = quad_rate / (2*math.pi*max_dev)
+        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.
         #
@@ -59,6 +59,7 @@ class wfm_rcv(gr.hier_block2):
                                               width_of_transition_band,
                                               fft.window.WIN_HAMMING)
         # input: float; output: float
-        self.audio_filter = filter.fir_filter_fff(audio_decimation, audio_coeffs)
+        self.audio_filter = filter.fir_filter_fff(
+            audio_decimation, audio_coeffs)
 
-        self.connect (self, self.fm_demod, self.audio_filter, self.deemph, self)
+        self.connect(self, self.fm_demod, self.audio_filter, self.deemph, self)
diff --git a/gr-analog/python/analog/wfm_rcv_fmdet.py b/gr-analog/python/analog/wfm_rcv_fmdet.py
index 3b587cc44c..d9289aa333 100644
--- a/gr-analog/python/analog/wfm_rcv_fmdet.py
+++ b/gr-analog/python/analog/wfm_rcv_fmdet.py
@@ -19,7 +19,7 @@ from .fm_emph import fm_deemph
 
 
 class wfm_rcv_fmdet(gr.hier_block2):
-    def __init__ (self, demod_rate, audio_decimation):
+    def __init__(self, demod_rate, audio_decimation):
         """
         Hierarchical block for demodulating a broadcast FM signal.
 
@@ -32,9 +32,10 @@ class wfm_rcv_fmdet(gr.hier_block2):
             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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(2, 2, gr.sizeof_float))      # Output signature
-        
+
         if audio_decimation != int(audio_decimation):
             raise ValueError("audio_decimation needs to be an integer")
         audio_decimation = int(audio_decimation)
@@ -51,26 +52,27 @@ class wfm_rcv_fmdet(gr.hier_block2):
         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_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
+        audio_coeffs = filter.firdes.low_pass(1.0,         # gain
                                               demod_rate,   # sampling rate
-                                              15000 ,
+                                              15000,
                                               width_of_transition_band,
                                               fft.window.WIN_HAMMING)
 
         # input: float; output: float
-        self.audio_filter = filter.fir_filter_fff(audio_decimation, audio_coeffs)
+        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
+            # NOTE THIS WAS HACKED TO OFFSET INSERTION LOSS DUE TO
+            # DEEMPHASIS
 
             stereo_carrier_filter_coeffs = \
                 filter.firdes.complex_band_pass(10.0,
@@ -80,9 +82,9 @@ class wfm_rcv_fmdet(gr.hier_block2):
                                                 width_of_transition_band,
                                                 fft.window.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
+            # 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
@@ -91,12 +93,12 @@ class wfm_rcv_fmdet(gr.hier_block2):
             stereo_dsbsc_filter_coeffs = \
                 filter.firdes.complex_band_pass(20.0,
                                                 demod_rate,
-                                                38000-15000 / 2,
-                                                38000+15000 / 2,
+                                                38000 - 15000 / 2,
+                                                38000 + 15000 / 2,
                                                 width_of_transition_band,
                                                 fft.window.WIN_HAMMING)
-            #print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs)
-            #print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs
+            # 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
@@ -106,7 +108,7 @@ class wfm_rcv_fmdet(gr.hier_block2):
 
             # carrier is twice the picked off carrier so arrange to do
             # a complex multiply
-            self.stereo_carrier_generator = blocks.multiply_cc();
+            self.stereo_carrier_generator = blocks.multiply_cc()
 
             # Pick off the rds signal
             stereo_rds_filter_coeffs = \
@@ -116,90 +118,96 @@ class wfm_rcv_fmdet(gr.hier_block2):
                                                 57000 + 1500,
                                                 width_of_transition_band,
                                                 fft.window.WIN_HAMMING)
-            #print "len stereo dsbsc filter = ",len(stereo_dsbsc_filter_coeffs)
-            #print "stereo dsbsc filter ", stereo_dsbsc_filter_coeffs
+            # 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);
+            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;
+            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);
+                                                                    min_freq)
 
-            #self.stereo_carrier_pll_recovery.squelch_enable(False)
-            ##pll_refout does not have squelch yet, so disabled for
-            #now
+            # 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();
+            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.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))
+                         (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))
+            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))
+            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))
+            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))
+            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))
+            # 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))
+            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))
+            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))
+            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)
-
+            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
@@ -208,14 +216,14 @@ class wfm_rcv_fmdet(gr.hier_block2):
 
             # 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))
+            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_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:
+        # else:
         #    self.connect (self.fm_demod, self.audio_filter, self)
diff --git a/gr-analog/python/analog/wfm_rcv_pll.py b/gr-analog/python/analog/wfm_rcv_pll.py
index 04ea9b1430..ce0e357d7d 100644
--- a/gr-analog/python/analog/wfm_rcv_pll.py
+++ b/gr-analog/python/analog/wfm_rcv_pll.py
@@ -16,6 +16,7 @@ from gnuradio import filter, fft
 from gnuradio.filter import firdes
 from gnuradio import analog
 
+
 class wfm_rcv_pll(gr.hier_block2):
     def __init__(self, demod_rate, audio_decimation, deemph_tau):
         """
@@ -30,7 +31,8 @@ class wfm_rcv_pll(gr.hier_block2):
             deemph_tau: deemphasis ime constant in seconds (75us in US, 50us in EUR). (float)
         """
         gr.hier_block2.__init__(self, "wfm_rcv_pll",
-                                gr.io_signature(1, 1, gr.sizeof_gr_complex), # Input signature
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_gr_complex),
                                 gr.io_signature(2, 2, gr.sizeof_float))      # Output signature
 
         if audio_decimation != int(audio_decimation):
@@ -42,56 +44,80 @@ class wfm_rcv_pll(gr.hier_block2):
         ##################################################
         self.demod_rate = demod_rate
         self.deemph_tau = deemph_tau
-        self.stereo_carrier_filter_coeffs = stereo_carrier_filter_coeffs = firdes.band_pass(-2.0, demod_rate, 37600, 38400, 400, fft.window.WIN_HAMMING, 6.76)
-        self.pilot_carrier_filter_coeffs = pilot_carrier_filter_coeffs = firdes.complex_band_pass(1.0, demod_rate, 18980, 19020, 1500, fft.window.WIN_HAMMING, 6.76)
+        self.stereo_carrier_filter_coeffs = stereo_carrier_filter_coeffs = firdes.band_pass(
+            -2.0, demod_rate, 37600, 38400, 400, fft.window.WIN_HAMMING, 6.76)
+        self.pilot_carrier_filter_coeffs = pilot_carrier_filter_coeffs = firdes.complex_band_pass(
+            1.0, demod_rate, 18980, 19020, 1500, fft.window.WIN_HAMMING, 6.76)
         self.deviation = deviation = 75000
-        self.audio_filter_coeffs = audio_filter_coeffs = firdes.low_pass(1, demod_rate, 15000,1500, fft.window.WIN_HAMMING, 6.76)
+        self.audio_filter_coeffs = audio_filter_coeffs = firdes.low_pass(
+            1, demod_rate, 15000, 1500, fft.window.WIN_HAMMING, 6.76)
         self.audio_decim = audio_decim = audio_decimation
         self.audio_rate = audio_rate = demod_rate / audio_decim
-        self.samp_delay = samp_delay = (len(pilot_carrier_filter_coeffs) - 1) // 2 + (len(stereo_carrier_filter_coeffs) - 1) // 2
+        self.samp_delay = samp_delay = (len(
+            pilot_carrier_filter_coeffs) - 1) // 2 + (len(stereo_carrier_filter_coeffs) - 1) // 2
 
         ##################################################
         # Blocks
         ##################################################
-        self.pilot_carrier_bpf = filter.fir_filter_fcc(1, pilot_carrier_filter_coeffs)
+        self.pilot_carrier_bpf = filter.fir_filter_fcc(
+            1, pilot_carrier_filter_coeffs)
         self.pilot_carrier_bpf.declare_sample_delay(0)
-        self.stereo_carrier_bpf = filter.fft_filter_fff(1, stereo_carrier_filter_coeffs, 1)
+        self.stereo_carrier_bpf = filter.fft_filter_fff(
+            1, stereo_carrier_filter_coeffs, 1)
         self.stereo_carrier_bpf.declare_sample_delay(0)
-        self.stereo_audio_lpf = filter.fft_filter_fff(audio_decim, audio_filter_coeffs, 1)
+        self.stereo_audio_lpf = filter.fft_filter_fff(
+            audio_decim, audio_filter_coeffs, 1)
         self.stereo_audio_lpf.declare_sample_delay(0)
-        self.mono_audio_lpf = filter.fft_filter_fff(audio_decim, audio_filter_coeffs, 1)
+        self.mono_audio_lpf = filter.fft_filter_fff(
+            audio_decim, audio_filter_coeffs, 1)
         self.mono_audio_lpf.declare_sample_delay(0)
         self.blocks_stereo_multiply = blocks.multiply_ff(1)
         self.blocks_pilot_multiply = blocks.multiply_cc(1)
         self.blocks_complex_to_imag = blocks.complex_to_imag(1)
         self.blocks_right_sub = blocks.sub_ff(1)
         self.blocks_left_add = blocks.add_ff(1)
-        self.analog_quadrature_demod_cf = analog.quadrature_demod_cf(demod_rate/(2*math.pi*deviation))
-        self.analog_pll_refout_cc = analog.pll_refout_cc(0.001, 2*math.pi * 19200 / demod_rate, 2*math.pi * 18800 / demod_rate)
-        self.analog_right_fm_deemph = analog.fm_deemph(fs=audio_rate, tau=deemph_tau)
-        self.analog_left_fm_deemph = analog.fm_deemph(fs=audio_rate, tau=deemph_tau)
-        self.blocks_delay_0 = blocks.delay(gr.sizeof_float*1, samp_delay)
+        self.analog_quadrature_demod_cf = analog.quadrature_demod_cf(
+            demod_rate / (2 * math.pi * deviation))
+        self.analog_pll_refout_cc = analog.pll_refout_cc(
+            0.001, 2 * math.pi * 19200 / demod_rate, 2 * math.pi * 18800 / demod_rate)
+        self.analog_right_fm_deemph = analog.fm_deemph(
+            fs=audio_rate, tau=deemph_tau)
+        self.analog_left_fm_deemph = analog.fm_deemph(
+            fs=audio_rate, tau=deemph_tau)
+        self.blocks_delay_0 = blocks.delay(gr.sizeof_float * 1, samp_delay)
 
         ##################################################
         # Connections
         ##################################################
         self.connect((self.analog_left_fm_deemph, 0), (self, 0))
         self.connect((self.analog_right_fm_deemph, 0), (self, 1))
-        self.connect((self.analog_pll_refout_cc, 0), (self.blocks_pilot_multiply, 1))
-        self.connect((self.analog_pll_refout_cc, 0), (self.blocks_pilot_multiply, 0))
-        self.connect((self.analog_quadrature_demod_cf, 0), (self.blocks_delay_0, 0))
-        self.connect((self.blocks_delay_0, 0), (self.blocks_stereo_multiply, 0))
+        self.connect((self.analog_pll_refout_cc, 0),
+                     (self.blocks_pilot_multiply, 1))
+        self.connect((self.analog_pll_refout_cc, 0),
+                     (self.blocks_pilot_multiply, 0))
+        self.connect((self.analog_quadrature_demod_cf, 0),
+                     (self.blocks_delay_0, 0))
+        self.connect((self.blocks_delay_0, 0),
+                     (self.blocks_stereo_multiply, 0))
         self.connect((self.blocks_delay_0, 0), (self.mono_audio_lpf, 0))
-        self.connect((self.analog_quadrature_demod_cf, 0), (self.pilot_carrier_bpf, 0))
-        self.connect((self.blocks_left_add, 0), (self.analog_left_fm_deemph, 0))
-        self.connect((self.blocks_right_sub, 0), (self.analog_right_fm_deemph, 0))
-        self.connect((self.blocks_complex_to_imag, 0), (self.stereo_carrier_bpf, 0))
-        self.connect((self.blocks_pilot_multiply, 0), (self.blocks_complex_to_imag, 0))
-        self.connect((self.blocks_stereo_multiply, 0), (self.stereo_audio_lpf, 0)) # L - R path
+        self.connect((self.analog_quadrature_demod_cf, 0),
+                     (self.pilot_carrier_bpf, 0))
+        self.connect((self.blocks_left_add, 0),
+                     (self.analog_left_fm_deemph, 0))
+        self.connect((self.blocks_right_sub, 0),
+                     (self.analog_right_fm_deemph, 0))
+        self.connect((self.blocks_complex_to_imag, 0),
+                     (self.stereo_carrier_bpf, 0))
+        self.connect((self.blocks_pilot_multiply, 0),
+                     (self.blocks_complex_to_imag, 0))
+        self.connect((self.blocks_stereo_multiply, 0),
+                     (self.stereo_audio_lpf, 0))  # L - R path
         self.connect((self.mono_audio_lpf, 0), (self.blocks_left_add, 1))
         self.connect((self.mono_audio_lpf, 0), (self.blocks_right_sub, 0))
         self.connect((self.stereo_audio_lpf, 0), (self.blocks_left_add, 0))
         self.connect((self.stereo_audio_lpf, 0), (self.blocks_right_sub, 1))
-        self.connect((self.stereo_carrier_bpf, 0), (self.blocks_stereo_multiply, 1))
-        self.connect((self.pilot_carrier_bpf, 0), (self.analog_pll_refout_cc, 0))
+        self.connect((self.stereo_carrier_bpf, 0),
+                     (self.blocks_stereo_multiply, 1))
+        self.connect((self.pilot_carrier_bpf, 0),
+                     (self.analog_pll_refout_cc, 0))
         self.connect((self, 0), (self.analog_quadrature_demod_cf, 0))
diff --git a/gr-analog/python/analog/wfm_tx.py b/gr-analog/python/analog/wfm_tx.py
index 11f90ae545..70e91f8261 100644
--- a/gr-analog/python/analog/wfm_tx.py
+++ b/gr-analog/python/analog/wfm_tx.py
@@ -35,16 +35,17 @@ class wfm_tx(gr.hier_block2):
         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
+                                # Input signature
+                                gr.io_signature(1, 1, gr.sizeof_float),
+                                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")
-
+            raise ValueError(
+                "quad_rate is not an integer multiple of audio_rate")
 
         do_interp = audio_rate != quad_rate
 
@@ -58,14 +59,16 @@ class wfm_tx(gr.hier_block2):
                                                  40)              # stopband atten dB
 
             print("len(interp_taps) =", len(interp_taps))
-            self.interpolator = filter.interp_fir_filter_fff (interp_factor, interp_taps)
+            self.interpolator = filter.interp_fir_filter_fff(
+                interp_factor, interp_taps)
 
         self.preemph = fm_preemph(quad_rate, tau=tau, fh=fh)
 
         k = 2 * math.pi * max_dev / quad_rate
-        self.modulator = analog.frequency_modulator_fc (k)
+        self.modulator = analog.frequency_modulator_fc(k)
 
         if do_interp:
-            self.connect(self, self.interpolator, self.preemph, self.modulator, self)
+            self.connect(self, self.interpolator,
+                         self.preemph, self.modulator, self)
         else:
             self.connect(self, self.preemph, self.modulator, self)
author	Josh Morman <jmorman@gnuradio.org>	2021-11-24 12:01:00 -0500
committer	mormj <34754695+mormj@users.noreply.github.com>	2021-11-24 14:41:53 -0500
commit	2ede969ec5f3f6d361814403b7375db5f15ec14d (patch)
tree	d963e92732842ee3491da1282dfa09400342c96d /gr-analog/python
parent	c6f291d57d913708a2517de3c6902425c2c575d3 (diff)