mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-24 09:10:36 +00:00
Code Issues 1 Projects Releases Wiki Activity

audiomixer: Replace racy timeout based tested with drain query

Browse source 
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 93d0b51dba
commit f98e457f99
1 changed files with 15 additions and 43 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

58
tests/check/elements/audiomixer.c
View file

@ -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;