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author | Sjoerd Simons <sjoerd@luon.net> | 2008-05-08 06:20:42 +0000 |
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committer | Sebastian Dröge <slomo@circular-chaos.org> | 2008-05-08 06:20:42 +0000 |
commit | 61aa25b04977de12f2a0793cf0341d47ee262faf (patch) | |
tree | b6b3519e850cb8e8c1d817f23db994ce5c1fc8a8 /tests | |
parent | cfe9a2f6bbdf51325cd1e9d715df0c655ed7f809 (diff) | |
download | gst-plugins-bad-61aa25b04977de12f2a0793cf0341d47ee262faf.tar.gz gst-plugins-bad-61aa25b04977de12f2a0793cf0341d47ee262faf.tar.bz2 gst-plugins-bad-61aa25b04977de12f2a0793cf0341d47ee262faf.zip |
gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
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.
Diffstat (limited to 'tests')
-rw-r--r-- | tests/check/elements/audioresample.c | 89 |
1 files changed, 89 insertions, 0 deletions
diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/audioresample.c index 4145be19..9481163f 100644 --- a/tests/check/elements/audioresample.c +++ b/tests/check/elements/audioresample.c @@ -414,6 +414,94 @@ GST_START_TEST (test_shutdown) 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 * audioresample_suite (void) { @@ -425,6 +513,7 @@ audioresample_suite (void) tcase_add_test (tc_chain, test_discont_stream); tcase_add_test (tc_chain, test_reuse); tcase_add_test (tc_chain, test_shutdown); + tcase_add_test (tc_chain, test_live_switch); return s; } |