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-rw-r--r--tests/check/Makefile.am2
-rw-r--r--tests/check/elements/legacyresample.c (renamed from tests/check/elements/audioresample.c)78
2 files changed, 40 insertions, 40 deletions
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 9d20a05c..45a5bc57 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -89,7 +89,7 @@ check_PROGRAMS = \
$(check_x264enc) \
elements/aacparse \
elements/amrparse \
- elements/audioresample \
+ elements/legacyresample \
elements/qtmux \
elements/selector \
elements/mxfdemux \
diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/legacyresample.c
index 598e4b12..4e804e3e 100644
--- a/tests/check/elements/audioresample.c
+++ b/tests/check/elements/legacyresample.c
@@ -1,6 +1,6 @@
/* GStreamer
*
- * unit test for audioresample
+ * unit test for legacyresample
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
@@ -52,14 +52,14 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
);
static GstElement *
-setup_audioresample (int channels, int inrate, int outrate)
+setup_legacyresample (int channels, int inrate, int outrate)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstCaps *caps;
GstStructure *structure;
- GST_DEBUG ("setup_audioresample");
- audioresample = gst_check_setup_element ("legacyresample");
+ GST_DEBUG ("setup_legacyresample");
+ legacyresample = gst_check_setup_element ("legacyresample");
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
structure = gst_caps_get_structure (caps, 0);
@@ -67,11 +67,11 @@ setup_audioresample (int channels, int inrate, int outrate)
"rate", G_TYPE_INT, inrate, NULL);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
"could not set to paused");
- mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
+ mysrcpad = gst_check_setup_src_pad (legacyresample, &srctemplate, caps);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
@@ -81,7 +81,7 @@ setup_audioresample (int channels, int inrate, int outrate)
"rate", G_TYPE_INT, outrate, NULL);
fail_unless (gst_caps_is_fixed (caps));
- mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
+ mysinkpad = gst_check_setup_sink_pad (legacyresample, &sinktemplate, caps);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_set_caps (mysinkpad, caps);
@@ -90,22 +90,22 @@ setup_audioresample (int channels, int inrate, int outrate)
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE);
- return audioresample;
+ return legacyresample;
}
static void
-cleanup_audioresample (GstElement * audioresample)
+cleanup_legacyresample (GstElement * legacyresample)
{
- GST_DEBUG ("cleanup_audioresample");
+ GST_DEBUG ("cleanup_legacyresample");
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
- gst_check_teardown_src_pad (audioresample);
- gst_check_teardown_sink_pad (audioresample);
- gst_check_teardown_element (audioresample);
+ gst_check_teardown_src_pad (legacyresample);
+ gst_check_teardown_sink_pad (legacyresample);
+ gst_check_teardown_element (legacyresample);
}
static void
@@ -145,7 +145,7 @@ static void
test_perfect_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
guint64 offset = 0;
@@ -153,11 +153,11 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
int i, j;
gint16 *p;
- audioresample = setup_audioresample (2, inrate, outrate);
+ legacyresample = setup_legacyresample (2, inrate, outrate);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
@@ -188,7 +188,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
fail_unless_equals_int (g_list_length (buffers), j);
}
- /* FIXME: we should make audioresample handle eos by flushing out the last
+ /* FIXME: we should make legacyresample handle eos by flushing out the last
* samples, which will give us one more, small, buffer */
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
@@ -197,7 +197,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
/* cleanup */
gst_caps_unref (caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
}
@@ -229,7 +229,7 @@ static void
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
@@ -240,11 +240,11 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
- audioresample = setup_audioresample (2, inrate, outrate);
+ legacyresample = setup_legacyresample (2, inrate, outrate);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
@@ -295,7 +295,7 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
/* cleanup */
gst_caps_unref (caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
}
GST_START_TEST (test_discont_stream)
@@ -321,16 +321,16 @@ GST_END_TEST;
GST_START_TEST (test_reuse)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstEvent *newseg;
GstBuffer *inbuffer;
GstCaps *caps;
- audioresample = setup_audioresample (1, 9343, 48000);
+ legacyresample = setup_legacyresample (1, 9343, 48000);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
@@ -351,10 +351,10 @@ GST_START_TEST (test_reuse)
fail_unless_equals_int (g_list_length (buffers), 1);
/* now reset and try again ... */
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
@@ -371,12 +371,12 @@ GST_START_TEST (test_reuse)
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... it also ends up being collected on the global buffer list. If we
- * now have more than 2 buffers, then audioresample probably didn't clean
+ * now have more than 2 buffers, then legacyresample probably didn't clean
* up its internal buffer properly and tried to push the remaining samples
* when it got the second NEWSEGMENT event */
fail_unless_equals_int (g_list_length (buffers), 2);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
gst_caps_unref (caps);
}
@@ -388,7 +388,7 @@ GST_START_TEST (test_shutdown)
GstCaps *caps;
guint i;
- /* create pipeline, force audioresample to actually resample */
+ /* create pipeline, force legacyresample to actually resample */
pipeline = gst_pipeline_new (NULL);
src = gst_check_setup_element ("audiotestsrc");
@@ -512,11 +512,11 @@ live_switch_push (int rate, GstCaps * caps)
GST_START_TEST (test_live_switch)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstEvent *newseg;
GstCaps *caps;
- audioresample = setup_audioresample (4, 48000, 48000);
+ legacyresample = setup_legacyresample (4, 48000, 48000);
/* Let the sinkpad act like something that can only handle things of
* rate 48000- and can only allocate buffers for that rate, but if someone
@@ -528,7 +528,7 @@ GST_START_TEST (test_live_switch)
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
@@ -545,14 +545,14 @@ GST_START_TEST (test_live_switch)
/* Downstream can provide the requested rate but will re-negotiate */
live_switch_push (50000, caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
gst_caps_unref (caps);
}
GST_END_TEST static Suite *
-audioresample_suite (void)
+legacyresample_suite (void)
{
- Suite *s = suite_create ("audioresample");
+ Suite *s = suite_create ("legacyresample");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
@@ -565,4 +565,4 @@ audioresample_suite (void)
return s;
}
-GST_CHECK_MAIN (audioresample);
+GST_CHECK_MAIN (legacyresample);
generated by cgit v1.2.1 (git 2.18.0) at 2024-08-06 14:37:59 +0000