16 files changed, 253 insertions, 62 deletions
diff --git a/docs/exploring-gnuradio/fm_tx.grc b/docs/exploring-gnuradio/fm_tx.grc
index 2f047bf09b..bb13417a6b 100644
--- a/docs/exploring-gnuradio/fm_tx.grc
+++ b/docs/exploring-gnuradio/fm_tx.grc
@@ -792,6 +792,10 @@
       <value>75e3</value>
     </param>
     <param>
+      <key>fh</key>
+      <value>0.925 * tx_rate/2.0</value>
+    </param>
+    <param>
       <key>affinity</key>
       <value></value>
     </param>
diff --git a/gr-analog/examples/fmtest.py b/gr-analog/examples/fmtest.py
index 327da8eacb..7ed08cafbe 100755
--- a/gr-analog/examples/fmtest.py
+++ b/gr-analog/examples/fmtest.py
@@ -48,7 +48,8 @@ class fmtx(gr.hier_block2):
                                 gr.io_signature(1, 1, gr.sizeof_float),
                                 gr.io_signature(1, 1, gr.sizeof_gr_complex))
 
-        fmtx = analog.nbfm_tx(audio_rate, if_rate, max_dev=5e3, tau=75e-6)
+        fmtx = analog.nbfm_tx(audio_rate, if_rate, max_dev=5e3,
+	                      tau=75e-6, fh=0.925*if_rate/2.0)
 
         # Local oscillator
         lo = analog.sig_source_c(if_rate,            # sample rate
diff --git a/gr-analog/grc/analog_fm_preemph.xml b/gr-analog/grc/analog_fm_preemph.xml
index fb898b87f3..a754ce9c6f 100644
--- a/gr-analog/grc/analog_fm_preemph.xml
+++ b/gr-analog/grc/analog_fm_preemph.xml
@@ -8,7 +8,7 @@
 	<name>FM Preemphasis</name>
 	<key>analog_fm_preemph</key>
 	<import>from gnuradio import analog</import>
-	<make>analog.fm_preemph(fs=$samp_rate, tau=$tau)</make>
+	<make>analog.fm_preemph(fs=$samp_rate, tau=$tau, fh=$fh)</make>
 	<param>
 		<name>Sample Rate</name>
 		<key>samp_rate</key>
@@ -20,6 +20,12 @@
 		<value>75e-6</value>
 		<type>real</type>
 	</param>
+	<param>
+		<name>High Corner Freq</name>
+		<key>fh</key>
+		<value>-1.0</value>
+		<type>real</type>
+	</param>
 	<sink>
 		<name>in</name>
 		<type>float</type>
diff --git a/gr-analog/grc/analog_nbfm_tx.xml b/gr-analog/grc/analog_nbfm_tx.xml
index ea13b8f1c9..bc80fffbcb 100644
--- a/gr-analog/grc/analog_nbfm_tx.xml
+++ b/gr-analog/grc/analog_nbfm_tx.xml
@@ -13,6 +13,7 @@
 	quad_rate=$quad_rate,
 	tau=$tau,
 	max_dev=$max_dev,
+	fh=$fh,
         )</make>
   <callback>set_max_deviation($max_dev)</callback>
 
@@ -42,6 +43,13 @@
     <type>real</type>
   </param>
 
+  <param>
+    <name>Preemphasis High Corner Freq</name>
+    <key>fh</key>
+    <value>-1.0</value>
+    <type>real</type>
+  </param>
+
   <check>($quad_rate)%($audio_rate) == 0</check>
 
   <sink>
diff --git a/gr-analog/grc/analog_wfm_tx.xml b/gr-analog/grc/analog_wfm_tx.xml
index 89844f9ef3..507c2ea894 100644
--- a/gr-analog/grc/analog_wfm_tx.xml
+++ b/gr-analog/grc/analog_wfm_tx.xml
@@ -13,6 +13,7 @@
 	quad_rate=$quad_rate,
 	tau=$tau,
 	max_dev=$max_dev,
+	fh=$fh,
 )</make>
 	<param>
 		<name>Audio Rate</name>
@@ -36,6 +37,12 @@
 		<value>75e3</value>
 		<type>real</type>
 	</param>
+	<param>
+		<name>Preemphasis High Corner Freq</name>
+		<key>fh</key>
+		<value>-1.0</value>
+		<type>real</type>
+	</param>
 	<check>($quad_rate)%($audio_rate) == 0</check>
 	<sink>
 		<name>in</name>
diff --git a/gr-analog/python/analog/fm_emph.py b/gr-analog/python/analog/fm_emph.py
index d2a38d4aff..bfa4742ace 100644
--- a/gr-analog/python/analog/fm_emph.py
+++ b/gr-analog/python/analog/fm_emph.py
@@ -21,17 +21,78 @@
 
 from gnuradio import gr, filter
 import math
+import cmath
 
 #
-#           1
-# H(s) = -------
-#         1 + s
+#  An analog deemphasis filter:
 #
-# tau is the RC time constant.
-# critical frequency: w_p = 1/tau
+#           R
+#  o------/\/\/\/---+----o
+#                   |
+#                   = C
+#                   |
+#                  ---
 #
-# We prewarp and use the bilinear z-transform to get our IIR coefficients.
-# See "Digital Signal Processing: A Practical Approach" by Ifeachor and Jervis
+#  Has this transfer function:
+#
+#             1            1
+#            ----         ---
+#             RC          tau
+#  H(s) = ---------- = ----------
+#               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 digitial 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
+#                 1 - -----         1 + -----
+#                      2 fs              2 fs
+#                               1 - ----------- z^-1
+#                                       -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
 #
 
 
@@ -51,15 +112,24 @@ class fm_deemph(gr.hier_block2):
                                 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
+        # Digital corner frequency
+        w_c = 1.0 / tau
+
+        # Prewarped analog corner frequency
+        w_ca = 2.0 * fs * math.tan(w_c / (2.0 * fs))
 
-        a1 = (w_pp - 1)/(w_pp + 1)
-        b0 = w_pp/(1 + w_pp)
-        b1 = b0
+        # Resulting digital pole, zero, and gain term from the bilinear
+        # transformation of H(s) = w_ca / (s + w_ca) to
+        # H(z) = b0 (1 - z1 z^-1)/(1 - p1 z^-1)
+        k = -w_ca / (2.0 * fs)
+        z1 = -1.0
+        p1 = (1.0 + k) / (1.0 - k)
+        b0 = -k / (1.0 - k)
 
-        btaps = [b0, b1]
-        ataps = [1, a1]
+        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
 
         if 0:
             print "btaps =", btaps
@@ -67,26 +137,17 @@ class fm_deemph(gr.hier_block2):
             global plot1
             plot1 = gru.gnuplot_freqz(gru.freqz(btaps, ataps), fs, True)
 
-        deemph = filter.iir_filter_ffd(btaps, ataps)
+        deemph = filter.iir_filter_ffd(btaps, ataps, False)
         self.connect(self, deemph, self)
 
 #
-#         1 + s*t1
-# H(s) = ----------
-#         1 + s*t2
-#
-# I think this is the right transfer function.
-#
+#  An analog preemphasis filter, that flattens out again at the high end:
 #
-# This fine ASCII rendition is based on Figure 5-15
-# in "Digital and Analog Communication Systems", Leon W. Couch II
-#
-#
-#               R1
+#               C
 #         +-----||------+
 #         |             |
 #  o------+             +-----+--------o
-#         |      C1     |     |
+#         |      R1     |     |
 #         +----/\/\/\/--+     \
 #                             /
 #                             \ R2
@@ -95,28 +156,94 @@ class fm_deemph(gr.hier_block2):
 #                             |
 #  o--------------------------+--------o
 #
-# f1 = 1/(2*pi*t1) = 1/(2*pi*R1*C)
+#  (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          R1 + R2
-# f2 = ------- = ------------
-#      2*pi*t2    2*pi*R1*R2*C
+#                   1
+#              s + ---
+#                  R1C
+#  H(s) = ------------------
+#               1       R1
+#          s + --- (1 + --)
+#              R1C      R2
 #
-# t1 is 75us in US, 50us in EUR
-# f2 should be higher than our audio bandwidth.
 #
+#  It has a corner due to the numerator, where the rise starts, at
 #
-# The Bode plot looks like this:
+#  |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
 #
-#                    /----------------
-#                   /
-#                  /  <-- slope = 20dB/decade
-#                 /
-#   -------------/
-#               f1    f2
+#  |Hn(j w_ch)|^2 = 1/2*|Hd(0)|^2  =>  s = j w_ch = j (1/(R1C) * (1 + R1/R2))
 #
-# We prewarp and use the bilinear z-transform to get our IIR coefficients.
-# See "Digital Signal Processing: A Practical Approach" by Ifeachor and Jervis
+#  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 digitial 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
+#                         1 - ------------ z^-1
+#              -w_cla              -w_cla
+#          1 - ------          1 - ------
+#               2 fs                2 fs
+#  H(z) = ------------ * -----------------------
+#              -w_cha              -w_cha
+#          1 - ------          1 + ------
+#               2 fs                2 fs
+#                         1 - ------------ z^-1
+#                                  -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
 #
 
 
@@ -124,21 +251,52 @@ class fm_preemph(gr.hier_block2):
     """
     FM Preemphasis IIR filter.
     """
-    def __init__(self, fs, tau=75e-6):
+    def __init__(self, fs, tau=75e-6, fh=-1.0):
         """
 
         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)
         """
-        gr.hier_block2.__init__(self, "fm_deemph",
+        gr.hier_block2.__init__(self, "fm_preemph",
                                 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
+	# Set fh to something sensible, if needed.
+	# N.B. fh == fs/2.0 or fh == 0.0 results in a pole on the unit circle
+	# 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
+
+	# Digital corner frequencies
+	w_cl = 1.0 / tau
+	w_ch = 2.0 * math.pi * fh
+
+	# Prewarped analog corner frequencies
+	w_cla = 2.0 * fs * math.tan(w_cl / (2.0 * fs))
+	w_cha = 2.0 * fs * math.tan(w_ch / (2.0 * fs))
+
+	# Resulting digital pole, zero, and gain term from the bilinear
+	# transformation of H(s) = (s + w_cla) / (s + w_cha) to
+	# H(z) = b0 (1 - z1 z^-1)/(1 - p1 z^-1)
+	kl = -w_cla / (2.0 * fs)
+	kh = -w_cha / (2.0 * fs)
+	z1 = (1.0 + kl) / (1.0 - kl)
+	p1 = (1.0 + kh) / (1.0 - kh)
+	b0 = (1.0 - kl) / (1.0 - kh)
+
+	# Since H(s = infinity) = 1.0, then H(z = -1) = 1.0 and
+	# this filter  has 0 dB gain at fs/2.0.
+	# 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))  \
+	   / (b0 * abs(1.0 - z1 * cmath.rect(1.0, -w_0dB)))
 
-        btaps = [1]
-        ataps = [1]
+	btaps = [ g * b0 * 1.0, g * b0 * -z1 ]
+	ataps = [          1.0,          -p1 ]
 
         if 0:
             print "btaps =", btaps
@@ -146,5 +304,5 @@ class fm_preemph(gr.hier_block2):
             global plot2
             plot2 = gru.gnuplot_freqz(gru.freqz(btaps, ataps), fs, True)
 
-        preemph = filter.iir_filter_ffd(btaps, ataps)
+        preemph = filter.iir_filter_ffd(btaps, ataps, False)
         self.connect(self, preemph, self)
diff --git a/gr-analog/python/analog/nbfm_tx.py b/gr-analog/python/analog/nbfm_tx.py
index ffd539ec55..aa6c1eccc7 100644
--- a/gr-analog/python/analog/nbfm_tx.py
+++ b/gr-analog/python/analog/nbfm_tx.py
@@ -29,7 +29,7 @@ except ImportError:
     import analog_swig as analog
 
 class nbfm_tx(gr.hier_block2):
-    def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=5e3):
+    def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=5e3, fh=-1.0):
         """
         Narrow Band FM Transmitter.
 
@@ -41,6 +41,7 @@ class nbfm_tx(gr.hier_block2):
             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)
+            fh: high frequency at which to flatten preemphasis; < 0 means default of 0.925*quad_rate/2.0 (float)
 
         quad_rate must be an integer multiple of audio_rate.
         """
@@ -71,7 +72,7 @@ class nbfm_tx(gr.hier_block2):
             #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)
+        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)
diff --git a/gr-analog/python/analog/wfm_tx.py b/gr-analog/python/analog/wfm_tx.py
index be662310db..a1b589350d 100644
--- a/gr-analog/python/analog/wfm_tx.py
+++ b/gr-analog/python/analog/wfm_tx.py
@@ -30,7 +30,7 @@ except ImportError:
     import analog_swig as analog
 
 class wfm_tx(gr.hier_block2):
-    def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=75e3):
+    def __init__(self, audio_rate, quad_rate, tau=75e-6, max_dev=75e3, fh=-1.0):
         """
         Wide Band FM Transmitter.
 
@@ -42,6 +42,7 @@ class wfm_tx(gr.hier_block2):
             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)
+            fh: high frequency at which to flatten preemphasis; < 0 means default of 0.925*quad_rate/2.0 (float)
 
         quad_rate must be an integer multiple of audio_rate.
         """
@@ -71,7 +72,7 @@ class wfm_tx(gr.hier_block2):
             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)
+        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)
diff --git a/gr-blocks/include/gnuradio/blocks/head.h b/gr-blocks/include/gnuradio/blocks/head.h
index 24772a43fa..a41137c023 100644
--- a/gr-blocks/include/gnuradio/blocks/head.h
+++ b/gr-blocks/include/gnuradio/blocks/head.h
@@ -47,7 +47,7 @@ namespace gr {
                        uint64_t nitems);
 
       virtual void reset() = 0;
-      virtual void set_length(int nitems) = 0;
+      virtual void set_length(uint64_t nitems) = 0;
     };
 
   } /* namespace blocks */
diff --git a/gr-blocks/lib/head_impl.h b/gr-blocks/lib/head_impl.h
index fe2918c404..4a2c33864b 100644
--- a/gr-blocks/lib/head_impl.h
+++ b/gr-blocks/lib/head_impl.h
@@ -39,7 +39,7 @@ namespace gr {
       ~head_impl();
 
       void reset() { d_ncopied_items = 0; }
-      void set_length(int nitems) { d_nitems = nitems; }
+      void set_length(uint64_t nitems) { d_nitems = nitems; }
 
       int work(int noutput_items,
                gr_vector_const_void_star &input_items,
diff --git a/gr-digital/lib/correlate_access_code_bb_ts_impl.cc b/gr-digital/lib/correlate_access_code_bb_ts_impl.cc
index c04bf4e180..e2fe37d8bb 100644
--- a/gr-digital/lib/correlate_access_code_bb_ts_impl.cc
+++ b/gr-digital/lib/correlate_access_code_bb_ts_impl.cc
@@ -84,8 +84,8 @@ namespace gr {
       if(d_len > 64)
         return false;
 
-      // set len top bits to 1.
-      d_mask = ((~0ULL) >> (64 - d_len)) << (64 - d_len);
+      // set len least significant bits to 1.
+      d_mask = ((~0ULL) >> (64 - d_len));
 
       d_access_code = 0;
       for(unsigned i=0; i < d_len; i++){
diff --git a/gr-digital/lib/correlate_access_code_ff_ts_impl.cc b/gr-digital/lib/correlate_access_code_ff_ts_impl.cc
index 86bbb686d5..dfbac40505 100644
--- a/gr-digital/lib/correlate_access_code_ff_ts_impl.cc
+++ b/gr-digital/lib/correlate_access_code_ff_ts_impl.cc
@@ -84,8 +84,8 @@ namespace gr {
       if(d_len > 64)
         return false;
 
-      // set len top bits to 1.
-      d_mask = ((~0ULL) >> (64 - d_len)) << (64 - d_len);
+      // set len least significant bits to 1.
+      d_mask = ((~0ULL) >> (64 - d_len));
 
       d_access_code = 0;
       for(unsigned i=0; i < d_len; i++){
diff --git a/gr-filter/examples/synth_to_chan.py b/gr-filter/examples/synth_to_chan.py
index 9e682021b7..88fb080a65 100755
--- a/gr-filter/examples/synth_to_chan.py
+++ b/gr-filter/examples/synth_to_chan.py
@@ -54,7 +54,7 @@ def main():
     fmtx = list()
     for fi in freqs:
         s = analog.sig_source_f(fs, analog.GR_SIN_WAVE, fi, 1)
-        fm = analog.nbfm_tx(fs, 4*fs, max_dev=10000, tau=75e-6)
+        fm = analog.nbfm_tx(fs, 4*fs, max_dev=10000, tau=75e-6, fh=0.925*(4*fs)/2.0)
         sigs.append(s)
         fmtx.append(fm)
 
diff --git a/gr-uhd/examples/python/fm_tx4.py b/gr-uhd/examples/python/fm_tx4.py
index fefa67861b..516033dae1 100755
--- a/gr-uhd/examples/python/fm_tx4.py
+++ b/gr-uhd/examples/python/fm_tx4.py
@@ -63,7 +63,8 @@ class pipeline(gr.hier_block2):
             sys.exit(1)
 
         print audio_rate, if_rate
-        fmtx = analog.nbfm_tx(audio_rate, if_rate, max_dev=5e3, tau=75e-6)
+        fmtx = analog.nbfm_tx(audio_rate, if_rate, max_dev=5e3,
+	                      tau=75e-6, fh=0.925*if_rate/2.0)
 
         # Local oscillator
         lo = analog.sig_source_c(if_rate,            # sample rate
diff --git a/gr-utils/python/modtool/gr-newmod/CMakeLists.txt b/gr-utils/python/modtool/gr-newmod/CMakeLists.txt
index 0513f44086..9b41b99f14 100644
--- a/gr-utils/python/modtool/gr-newmod/CMakeLists.txt
+++ b/gr-utils/python/modtool/gr-newmod/CMakeLists.txt
@@ -112,6 +112,7 @@ find_package(Doxygen)
 # API compatible version required.
 set(GR_REQUIRED_COMPONENTS RUNTIME)
 find_package(Gnuradio "3.7.2" REQUIRED)
+list(INSERT CMAKE_MODULE_PATH 0 ${CMAKE_SOURCE_DIR}/cmake/Modules)
 
 if(NOT CPPUNIT_FOUND)
     message(FATAL_ERROR "CppUnit required to compile howto")
diff --git a/gr-utils/python/modtool/gr-newmod/cmake/Modules/GrMiscUtils.cmake b/gr-utils/python/modtool/gr-newmod/cmake/Modules/GrMiscUtils.cmake
index 188c40480b..5bad57c51e 100644
--- a/gr-utils/python/modtool/gr-newmod/cmake/Modules/GrMiscUtils.cmake
+++ b/gr-utils/python/modtool/gr-newmod/cmake/Modules/GrMiscUtils.cmake
@@ -363,6 +363,7 @@ macro(GR_EXPAND_X_H component root)
 
 import sys, os, re
 sys.path.append('${GR_RUNTIME_PYTHONPATH}')
+sys.path.append('${CMAKE_SOURCE_DIR}/python')
 os.environ['srcdir'] = '${CMAKE_CURRENT_SOURCE_DIR}'
 os.chdir('${CMAKE_CURRENT_BINARY_DIR}')
 
@@ -406,6 +407,7 @@ macro(GR_EXPAND_X_CC_H component root)
 
 import sys, os, re
 sys.path.append('${GR_RUNTIME_PYTHONPATH}')
+sys.path.append('${CMAKE_SOURCE_DIR}/python')
 os.environ['srcdir'] = '${CMAKE_CURRENT_SOURCE_DIR}'
 os.chdir('${CMAKE_CURRENT_BINARY_DIR}')
 
@@ -466,6 +468,7 @@ macro(GR_EXPAND_X_CC_H_IMPL component root)
 
 import sys, os, re
 sys.path.append('${GR_RUNTIME_PYTHONPATH}')
+sys.path.append('${CMAKE_SOURCE_DIR}/python')
 os.environ['srcdir'] = '${CMAKE_CURRENT_SOURCE_DIR}'
 os.chdir('${CMAKE_CURRENT_BINARY_DIR}')