[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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