[gst-cvs] gst-plugins-base: check: Improve audioresample test

Jan Schmidt thaytan at kemper.freedesktop.org
Fri Sep 11 13:45:17 PDT 2009


Module: gst-plugins-base
Branch: master
Commit: a9f815bd8de54cdb4b063121cba7592b80289a30
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=a9f815bd8de54cdb4b063121cba7592b80289a30

Author: Jan Schmidt <thaytan at noraisin.net>
Date:   Fri Sep 11 21:44:18 2009 +0100

check: Improve audioresample test

Make the audioresample test work with CK_FORK=no, and
turn a g_print into a GST_INFO.

---

 tests/check/elements/audioresample.c |    3 ++-
 1 files changed, 2 insertions(+), 1 deletions(-)

diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/audioresample.c
index b9eb520..4a67a9e 100644
--- a/tests/check/elements/audioresample.c
+++ b/tests/check/elements/audioresample.c
@@ -131,6 +131,7 @@ cleanup_audioresample (GstElement * audioresample)
   gst_check_teardown_src_pad (audioresample);
   gst_check_teardown_sink_pad (audioresample);
   gst_check_teardown_element (audioresample);
+  gst_check_drop_buffers ();
 }
 
 static void
@@ -792,7 +793,7 @@ fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
   }
 
   if (ctx->in_buffer_count != ctx->out_buffer_count) {
-    g_print ("timestamp %" GST_TIME_FORMAT "\n",
+    GST_INFO ("timestamp %" GST_TIME_FORMAT,
         GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
   }
 





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