diff --git a/webrtc/modules/audio_processing/test/py_conversational_speech/README.md b/webrtc/modules/audio_processing/test/py_conversational_speech/README.md
index 432448e278..79d07fdf08 100644
--- a/webrtc/modules/audio_processing/test/py_conversational_speech/README.md
+++ b/webrtc/modules/audio_processing/test/py_conversational_speech/README.md
@@ -26,12 +26,16 @@ IMPORTANT: **the whole code has not been landed yet.**
 For each end, there is a set of audio tracks, e.g., a1, a2 and a3 (speaker A)
 and b1, b2 (speaker B).
 The text file with the timing information may look like this:
-```  A a1 0
-  B b1 0
-  A a2 100
-  B b2 -200
-  A a3 0
-  A a4 0```
+
+```
+A a1 0
+B b1 0
+A a2 100
+B b2 -200
+A a3 0
+A a4 0
+```
+
 The first column indicates the speaker name, the second contains the audio track
 file names, and the third the offsets (in milliseconds) used to concatenate the
 chunks.
@@ -39,26 +43,32 @@ chunks.
 Assume that all the audio tracks in the example above are 1000 ms long.
 The tool will then generate two tracks (A and B) that look like this:
 
-```Track A:
+**Track A**
+```
   a1 (1000 ms)
   silence (1100 ms)
   a2 (1000 ms)
   silence (800 ms)
   a3 (1000 ms)
-  a4 (1000 ms)```
+  a4 (1000 ms)
+```
 
-```Track B:
+**Track B**
+```
   silence (1000 ms)
   b1 (1000 ms)
   silence (900 ms)
   b2 (1000 ms)
-  silence (2000 ms)```
+  silence (2000 ms)
+```
 
 The two tracks can be also visualized as follows (one characheter represents
 100 ms, "." is silence and "*" is speech).
 
-```t: 0         1        2        3        4        5        6 (s)
+```
+t: 0         1        2        3        4        5        6 (s)
 A: **********...........**********........********************
 B: ..........**********.........**********....................
                                 ^ 200 ms cross-talk
-        100 ms silence ^```
+        100 ms silence ^
+```
diff --git a/webrtc/modules/audio_processing/test/py_conversational_speech/generate_conversational_tracks.py b/webrtc/modules/audio_processing/test/py_conversational_speech/generate_conversational_tracks.py
new file mode 100644
index 0000000000..41e3bbdc6b
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/py_conversational_speech/generate_conversational_tracks.py
@@ -0,0 +1,58 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Generate multiple-end audio tracks to simulate conversational
+   speech with two or more participants.
+
+Usage: generate_conversational_tracks.py
+          -i path/to/source/audiotracks
+          -t path/to/timing_file.txt
+          -o output/path
+"""
+
+import argparse
+import logging
+import sys
+
+def _InstanceArgumentsParser():
+  parser = argparse.ArgumentParser(description=(
+      'Generate multiple-end audio tracks to simulate conversational speech '
+      'with two or more participants.'))
+
+  parser.add_argument('-i', '--input_tracks_path', required=True,
+                      help='directory containing the speech turn wav files')
+
+  parser.add_argument('-t', '--timing_file', required=True,
+                      help='path to the timing text file')
+
+  parser.add_argument('-o', '--output_dir', required=False,
+                      help=('base path to the output directory in which the '
+                            'output wav files are saved'),
+                      default='output')
+
+  return parser
+
+
+def main():
+  # TODO(alessiob): level = logging.INFO once debugged.
+  logging.basicConfig(level=logging.DEBUG)
+
+  parser = _InstanceArgumentsParser()
+  args = parser.parse_args()
+
+  # TODO(alessiob): pass the arguments to the app controller.
+
+  # TODO(alessiob): remove when comment above addressed.
+  logging.debug(args)
+
+  sys.exit(0)
+
+
+if __name__ == '__main__':
+  main()
diff --git a/webrtc/modules/audio_processing/test/py_conversational_speech/test_generation.py b/webrtc/modules/audio_processing/test/py_conversational_speech/test_generation.py
new file mode 100644
index 0000000000..3fc34e399d
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/py_conversational_speech/test_generation.py
@@ -0,0 +1,20 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import unittest
+
+import generate_conversational_tracks
+
+class TestGenerationScript(unittest.TestCase):
+
+  def TestMain(self):
+    # Exit with error code if no arguments are passed.
+    with self.assertRaises(SystemExit) as cm:
+      generate_conversational_tracks.main()
+    self.assertGreater(cm.exception.code, 0)