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audiomixer: Replace racy timeout based tested with drain query

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Using the drain query, we can be certain that the buffer has done going
through the aggregator by taking the stream locks.

https://bugzilla.gnome.org/show_bug.cgi?id=742684
This commit is contained in:
Olivier Crête 2015-01-22 17:41:24 -05:00 committed by Thibault Saunier
parent 402c0d4c5c
commit 0955a39a3d
1 changed files with 15 additions and 43 deletions
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58
tests/check/elements/audiomixer.c
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@ -891,36 +891,17 @@ GST_END_TEST;
static GstBuffer *handoff_buffer = NULL;
static gboolean
_quit (GMainLoop * ml)
{
g_main_loop_quit (ml);
return G_SOURCE_REMOVE;
}
static void
handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
GstClockTime * wanted_end)
gpointer user_data)
{
GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
" -- %p DURATION is %" GST_TIME_FORMAT " -- WANTED END %" GST_TIME_FORMAT,
" -- %p DURATION is %" GST_TIME_FORMAT,
gst_buffer_get_size (buffer), buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)),
GST_TIME_ARGS (*wanted_end));
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
gst_buffer_replace (&handoff_buffer, buffer);
/* Buffers we push in will be 'cut' into different smaller buffers,
* we make sure that the last chunck was pushes before we concider the buffer
* we pushed as being done */
if (main_loop && *wanted_end
&& *wanted_end <=
GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)) {
*wanted_end = 0;
g_idle_add ((GSourceFunc) _quit, main_loop);
}
}
/* check if clipping works as expected */
@ -936,14 +917,10 @@ GST_START_TEST (test_clip)
GstEvent *event;
GstBuffer *buffer;
GstCaps *caps;
GMainLoop *local_mainloop;
GstClockTime wanted_end = 0;
GstQuery *drain = gst_query_new_drain ();
GST_INFO ("preparing test");
local_mainloop = g_main_loop_new (NULL, FALSE);
main_loop = NULL;
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
@ -957,8 +934,7 @@ GST_START_TEST (test_clip)
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb,
&wanted_end);
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
res = gst_element_link (audiomixer, sink);
@ -999,16 +975,13 @@ GST_START_TEST (test_clip)
buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
ret = gst_pad_chain (sinkpad, buffer);
main_loop = local_mainloop;
ck_assert_int_eq (ret, GST_FLOW_OK);
/* The aggregation is done in a dedicated thread, so we can't
* not know when it is actually going to happen, so we just add\
* a 100 ms timeout to be able to then check that the aggregation
* did happen as we do not have much other choice.
* not know when it is actually going to happen, so we use a DRAIN query
* to wait for it to complete.
*/
g_timeout_add (100, (GSourceFunc) _quit, main_loop);
g_main_loop_run (main_loop);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer == NULL);
/* should be partially clipped */
@ -1016,15 +989,14 @@ GST_START_TEST (test_clip)
GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
wanted_end = 135 * GST_MSECOND;
GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
main_loop = local_mainloop;
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
g_main_loop_run (main_loop);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer != NULL);
gst_buffer_replace (&handoff_buffer, NULL);
@ -1033,18 +1005,18 @@ GST_START_TEST (test_clip)
GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
wanted_end = 390 * GST_MSECOND;
GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
g_main_loop_run (main_loop);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer != NULL);
gst_buffer_replace (&handoff_buffer, NULL);
fail_unless (handoff_buffer == NULL);
/* should be clipped and ok */
buffer = gst_buffer_new_and_alloc (44100);
GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
@ -1052,9 +1024,8 @@ GST_START_TEST (test_clip)
buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
ret = gst_pad_chain (sinkpad, buffer);
g_timeout_add (100, (GSourceFunc) _quit, main_loop);
g_main_loop_run (main_loop);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer == NULL);
gst_element_release_request_pad (audiomixer, sinkpad);
@ -1063,6 +1034,7 @@ GST_START_TEST (test_clip)
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
gst_query_unref (drain);
}
GST_END_TEST;