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gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...

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Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
This commit is contained in:
Sjoerd Simons 2008-05-08 06:20:42 +00:00 committed by Sebastian Dröge
parent cfe9a2f6bb
commit 61aa25b049
2 changed files with 89 additions and 4 deletions
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4
gst/audioresample/gstaudioresample.c
View file

@ -192,10 +192,6 @@ gst_audioresample_init (GstAudioresample * audioresample,
trans = GST_BASE_TRANSFORM (audioresample); trans = GST_BASE_TRANSFORM (audioresample);
/* buffer alloc passthrough is too impossible. FIXME, it
* is trivial in the passthrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
audioresample->filter_length = DEFAULT_FILTERLEN; audioresample->filter_length = DEFAULT_FILTERLEN;
audioresample->need_discont = FALSE; audioresample->need_discont = FALSE;

89
tests/check/elements/audioresample.c
View file

@ -414,6 +414,94 @@ GST_START_TEST (test_shutdown)
gst_object_unref (pipeline); gst_object_unref (pipeline);
} }
GST_END_TEST;
static GstFlowReturn
alloc_only_48000 (GstPad * pad, guint64 offset, guint size, GstCaps * caps,
GstBuffer ** buf)
{
GstStructure *structure;
gint rate;
structure = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (structure, "rate", &rate));
if (rate != 48000)
return GST_FLOW_NOT_NEGOTIATED;
*buf = NULL;
return GST_FLOW_OK;
}
GST_START_TEST (test_live_switch)
{
GstElement *audioresample;
GstEvent *newseg;
GstBuffer *inbuffer;
GstCaps *caps;
GstCaps *newcaps;
GList *l;
audioresample = setup_audioresample (1, 48000, 48000);
/* Let the sinkpad act like something that can only handle things of
* rate 48000 and can only allocate buffers for that rate */
gst_pad_set_bufferalloc_function (mysinkpad, alloc_only_48000);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, 48000 * 4, caps, &inbuffer) == GST_FLOW_OK);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, caps);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
/* Prepare a new buffer, but now with different caps */
fail_unless ((newcaps =
gst_caps_make_writable (gst_caps_ref (caps))) != NULL);
gst_caps_set_simple (newcaps, "rate", G_TYPE_INT, 1234, NULL);
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, 1234 * 4, newcaps, &inbuffer) == GST_FLOW_OK);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, newcaps);
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless_equals_int (g_list_length (buffers), 2);
cleanup_audioresample (audioresample);
for (l = buffers; l; l = l->next) {
GstBuffer *buffer = GST_BUFFER (l->data);
gst_buffer_unref (buffer);
}
g_list_free (buffers);
buffers = NULL;
gst_caps_unref (caps);
gst_caps_unref (newcaps);
}
GST_END_TEST static Suite * GST_END_TEST static Suite *
audioresample_suite (void) audioresample_suite (void)
{ {
@ -425,6 +513,7 @@ audioresample_suite (void)
tcase_add_test (tc_chain, test_discont_stream); tcase_add_test (tc_chain, test_discont_stream);
tcase_add_test (tc_chain, test_reuse); tcase_add_test (tc_chain, test_reuse);
tcase_add_test (tc_chain, test_shutdown); tcase_add_test (tc_chain, test_shutdown);
tcase_add_test (tc_chain, test_live_switch);
return s; return s;
} }