[gst-cvs] gst-python: Add test_audio.py.

[Alessandro Decina]

Module: gst-python
Branch: master
Commit: 7c6041d4795bfdd6aaab9ebefd4ed8ce67197969
URL:    http://cgit.freedesktop.org/gstreamer/gst-python/commit/?id=7c6041d4795bfdd6aaab9ebefd4ed8ce67197969

Author: Edward Hervey <bilboed at bilboed.com>
Date:   Fri Aug 21 16:41:29 2009 +0200

Add test_audio.py.

---

 testsuite/Makefile.am   |    1 +
 testsuite/test_audio.py |   38 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 39 insertions(+), 0 deletions(-)

diff --git a/testsuite/Makefile.am b/testsuite/Makefile.am
index 552ca3b..282e362 100644
--- a/testsuite/Makefile.am
+++ b/testsuite/Makefile.am
@@ -20,6 +20,7 @@ testhelper.la: $(testhelper_la_OBJECTS) $(testhelper_la_DEPENDENCIES)
 
 tests =  \
 	test_adapter.py \
+	test_audio.py \
 	test_bin.py \
 	test_buffer.py \
         test_caps.py \
diff --git a/testsuite/test_audio.py b/testsuite/test_audio.py
new file mode 100644
index 0000000..60321c8
--- /dev/null
+++ b/testsuite/test_audio.py
@@ -0,0 +1,38 @@
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+#
+# gst-python - Python bindings for GStreamer
+# Copyright (C) 2009 Edward Hervey <edward.hervey at collabora.co.uk>
+#
+# This library is free software; you can redistribute it and/or
+# modify it under the terms of the GNU Lesser General Public
+# License as published by the Free Software Foundation; either
+# version 2.1 of the License, or (at your option) any later version.
+#
+# This library 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
+# Lesser General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public
+# License along with this library; if not, write to the Free Software
+# Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA
+
+from common import gobject, gst, unittest, TestCase
+
+class Audio(TestCase):
+
+    def testBufferclip(self):
+        assert hasattr(gst.audio, "buffer_clip")
+        # create a segment
+        segment = gst.Segment()
+        gst.debug("Created the new segment")
+        # we'll put a new segment of 500ms to 1000ms
+        segment.set_newsegment(False, 1.0, gst.FORMAT_TIME, 0, -1, 0)
+        gst.debug("Initialized the new segment")
+        # create a new dummy buffer
+        b = gst.Buffer("this is a really useless line")
+        gst.debug("Created the buffer")
+        # clip... which shouldn't do anything
+        b2 = gst.audio.buffer_clip(b, segment, 44100, 8)
+        gst.debug("DONE !")