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gstreamer/tests/check/elements/audiomixer.c
Nirbheek Chauhan ad8cb458ba audioaggregator: Sync pad values before aggregating 
We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.

Also add a test for this.

https://bugzilla.gnome.org/show_bug.cgi?id=749574
2015-07-22 19:50:38 +01:00

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/* GStreamer
*
* unit test for audiomixer
*
* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#ifdef HAVE_VALGRIND
# include <valgrind/valgrind.h>
#endif
#include <unistd.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gstconsistencychecker.h>
#include <gst/audio/audio.h>
#include <gst/base/gstbasesrc.h>
#include <gst/controller/gstdirectcontrolbinding.h>
#include <gst/controller/gstinterpolationcontrolsource.h>
static GMainLoop *main_loop;
/* make sure downstream gets a CAPS event before buffers are sent */
GST_START_TEST (test_caps)
{
GstElement *pipeline, *src, *audiomixer, *sink;
GstStateChangeReturn state_res;
GstCaps *caps;
GstPad *pad;
/* build pipeline */
pipeline = gst_pipeline_new ("pipeline");
src = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (pipeline), src, audiomixer, sink, NULL);
fail_unless (gst_element_link_many (src, audiomixer, sink, NULL));
/* prepare playing */
state_res = gst_element_set_state (pipeline, GST_STATE_PAUSED);
fail_unless_equals_int (state_res, GST_STATE_CHANGE_ASYNC);
/* wait for preroll */
state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
fail_unless_equals_int (state_res, GST_STATE_CHANGE_SUCCESS);
/* check caps on fakesink */
pad = gst_element_get_static_pad (sink, "sink");
caps = gst_pad_get_current_caps (pad);
fail_unless (caps != NULL);
gst_caps_unref (caps);
gst_object_unref (pad);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
/* check that caps set on the property are honoured */
GST_START_TEST (test_filter_caps)
{
GstElement *pipeline, *src, *audiomixer, *sink;
GstStateChangeReturn state_res;
GstCaps *filter_caps, *caps;
GstPad *pad;
filter_caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
"channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL);
/* build pipeline */
pipeline = gst_pipeline_new ("pipeline");
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (src, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", NULL);
g_object_set (audiomixer, "caps", filter_caps, NULL);
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (pipeline), src, audiomixer, sink, NULL);
fail_unless (gst_element_link_many (src, audiomixer, sink, NULL));
/* prepare playing */
state_res = gst_element_set_state (pipeline, GST_STATE_PAUSED);
fail_unless_equals_int (state_res, GST_STATE_CHANGE_ASYNC);
/* wait for preroll */
state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
fail_unless_equals_int (state_res, GST_STATE_CHANGE_SUCCESS);
/* check caps on fakesink */
pad = gst_element_get_static_pad (sink, "sink");
caps = gst_pad_get_current_caps (pad);
fail_unless (caps != NULL);
GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps);
fail_unless (gst_caps_is_equal_fixed (caps, filter_caps));
gst_caps_unref (caps);
gst_object_unref (pad);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
gst_caps_unref (filter_caps);
}
GST_END_TEST;
static gboolean
set_playing (GstElement * element)
{
GstStateChangeReturn state_res;
state_res = gst_element_set_state (element, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
return FALSE;
}
static void
message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
{
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
GST_MESSAGE_SRC (message), message);
switch (message->type) {
case GST_MESSAGE_EOS:
g_main_loop_quit (main_loop);
break;
case GST_MESSAGE_WARNING:{
GError *gerror;
gchar *debug;
gst_message_parse_warning (message, &gerror, &debug);
gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
g_error_free (gerror);
g_free (debug);
break;
}
case GST_MESSAGE_ERROR:{
GError *gerror;
gchar *debug;
gst_message_parse_error (message, &gerror, &debug);
gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
g_error_free (gerror);
g_free (debug);
g_main_loop_quit (main_loop);
break;
}
default:
break;
}
}
static GstFormat format = GST_FORMAT_UNDEFINED;
static gint64 position = -1;
static void
test_event_message_received (GstBus * bus, GstMessage * message,
GstPipeline * bin)
{
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
GST_MESSAGE_SRC (message), message);
switch (message->type) {
case GST_MESSAGE_SEGMENT_DONE:
gst_message_parse_segment_done (message, &format, &position);
GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
g_main_loop_quit (main_loop);
break;
default:
g_assert_not_reached ();
break;
}
}
GST_START_TEST (test_event)
{
GstElement *bin, *src1, *src2, *audiomixer, *sink;
GstBus *bus;
GstEvent *seek_event;
GstStateChangeReturn state_res;
gboolean res;
GstPad *srcpad, *sinkpad;
GstStreamConsistency *chk_1, *chk_2, *chk_3;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
res = gst_element_link (src1, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (src2, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
srcpad = gst_element_get_static_pad (audiomixer, "src");
chk_3 = gst_consistency_checker_new (srcpad);
gst_object_unref (srcpad);
/* create consistency checkers for the pads */
srcpad = gst_element_get_static_pad (src1, "src");
chk_1 = gst_consistency_checker_new (srcpad);
sinkpad = gst_pad_get_peer (srcpad);
gst_consistency_checker_add_pad (chk_3, sinkpad);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
srcpad = gst_element_get_static_pad (src2, "src");
chk_2 = gst_consistency_checker_new (srcpad);
sinkpad = gst_pad_get_peer (srcpad);
gst_consistency_checker_add_pad (chk_3, sinkpad);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
format = GST_FORMAT_UNDEFINED;
position = -1;
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done",
(GCallback) test_event_message_received, bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
GST_INFO ("starting test");
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (bin, seek_event);
fail_unless (res == TRUE, NULL);
/* run pipeline */
g_idle_add ((GSourceFunc) set_playing, bin);
GST_INFO ("running main loop");
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
ck_assert_int_eq (position, 2 * GST_SECOND);
/* cleanup */
g_main_loop_unref (main_loop);
gst_consistency_checker_free (chk_1);
gst_consistency_checker_free (chk_2);
gst_consistency_checker_free (chk_3);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
static guint play_count = 0;
static GstEvent *play_seek_event = NULL;
static void
test_play_twice_message_received (GstBus * bus, GstMessage * message,
GstPipeline * bin)
{
gboolean res;
GstStateChangeReturn state_res;
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
GST_MESSAGE_SRC (message), message);
switch (message->type) {
case GST_MESSAGE_SEGMENT_DONE:
play_count++;
if (play_count == 1) {
state_res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_READY);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* prepare playing again */
state_res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (GST_ELEMENT (bin),
gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
state_res =
gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
} else {
g_main_loop_quit (main_loop);
}
break;
default:
g_assert_not_reached ();
break;
}
}
GST_START_TEST (test_play_twice)
{
GstElement *bin, *src1, *src2, *audiomixer, *sink;
GstBus *bus;
gboolean res;
GstStateChangeReturn state_res;
GstPad *srcpad;
GstStreamConsistency *consist;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
res = gst_element_link (src1, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (src2, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
srcpad = gst_element_get_static_pad (audiomixer, "src");
consist = gst_consistency_checker_new (srcpad);
gst_object_unref (srcpad);
play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
play_count = 0;
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done",
(GCallback) test_play_twice_message_received, bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
GST_INFO ("starting test");
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
g_idle_add ((GSourceFunc) set_playing, bin);
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
ck_assert_int_eq (play_count, 2);
/* cleanup */
g_main_loop_unref (main_loop);
gst_consistency_checker_free (consist);
gst_event_ref (play_seek_event);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
GST_START_TEST (test_play_twice_then_add_and_play_again)
{
GstElement *bin, *src1, *src2, *src3, *audiomixer, *sink;
GstBus *bus;
gboolean res;
GstStateChangeReturn state_res;
gint i;
GstPad *srcpad;
GstStreamConsistency *consist;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
srcpad = gst_element_get_static_pad (audiomixer, "src");
consist = gst_consistency_checker_new (srcpad);
gst_object_unref (srcpad);
res = gst_element_link (src1, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (src2, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done",
(GCallback) test_play_twice_message_received, bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
/* run it twice */
for (i = 0; i < 2; i++) {
play_count = 0;
GST_INFO ("starting test-loop %d", i);
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
g_idle_add ((GSourceFunc) set_playing, bin);
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_READY);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
ck_assert_int_eq (play_count, 2);
/* plug another source */
if (i == 0) {
src3 = gst_element_factory_make ("audiotestsrc", "src3");
g_object_set (src3, "wave", 4, NULL); /* silence */
gst_bin_add (GST_BIN (bin), src3);
res = gst_element_link (src3, audiomixer);
fail_unless (res == TRUE, NULL);
}
gst_consistency_checker_reset (consist);
}
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* cleanup */
g_main_loop_unref (main_loop);
gst_event_ref (play_seek_event);
gst_consistency_checker_free (consist);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
static void
test_live_seeking_eos_message_received (GstBus * bus, GstMessage * message,
GstPipeline * bin)
{
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
GST_MESSAGE_SRC (message), message);
switch (message->type) {
case GST_MESSAGE_EOS:
g_main_loop_quit (main_loop);
break;
default:
g_assert_not_reached ();
break;
}
}
static GstElement *
test_live_seeking_try_audiosrc (const gchar * factory_name)
{
GstElement *src;
GstStateChangeReturn state_res;
if (!(src = gst_element_factory_make (factory_name, NULL))) {
GST_INFO ("can't make '%s', skipping", factory_name);
return NULL;
}
/* Test that the audio source can get to ready, else skip */
state_res = gst_element_set_state (src, GST_STATE_READY);
gst_element_set_state (src, GST_STATE_NULL);
if (state_res == GST_STATE_CHANGE_FAILURE) {
GST_INFO_OBJECT (src, "can't go to ready, skipping");
gst_object_unref (src);
return NULL;
}
return src;
}
/* test failing seeks on live-sources */
GST_START_TEST (test_live_seeking)
{
GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink;
GstCaps *caps;
GstBus *bus;
gboolean res;
GstPad *srcpad;
GstPad *sinkpad;
gint i;
GstStateChangeReturn state_res;
GstStreamConsistency *consist;
/* don't use autoaudiosrc, as then we can't set anything here */
const gchar *audio_src_factories[] = {
"alsasrc",
"pulseaudiosrc"
};
GST_INFO ("preparing test");
main_loop = NULL;
play_seek_event = NULL;
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) {
src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]);
}
if (!src1) {
/* normal audiosources behave differently than audiotestsrc */
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, "is-live", TRUE, NULL); /* silence */
} else {
/* live sources ignore seeks, force eos after 2 sec (4 buffers half second
* each)
*/
g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL);
}
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
cf = gst_element_factory_make ("capsfilter", "capsfilter");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL);
res = gst_element_link (src1, cf);
fail_unless (res == TRUE, NULL);
res = gst_element_link (cf, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
gst_element_set_state (bin, GST_STATE_PLAYING);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
sinkpad = gst_element_get_static_pad (sink, "sink");
fail_unless (sinkpad != NULL);
caps = gst_pad_get_current_caps (sinkpad);
fail_unless (caps != NULL);
gst_object_unref (sinkpad);
gst_element_set_state (bin, GST_STATE_NULL);
g_object_set (cf, "caps", caps, NULL);
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
gst_bin_add (GST_BIN (bin), src2);
res = gst_element_link_filtered (src2, audiomixer, caps);
fail_unless (res == TRUE, NULL);
gst_caps_unref (caps);
play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos",
(GCallback) test_live_seeking_eos_message_received, bin);
srcpad = gst_element_get_static_pad (audiomixer, "src");
consist = gst_consistency_checker_new (srcpad);
gst_object_unref (srcpad);
GST_INFO ("starting test");
/* run it twice */
for (i = 0; i < 2; i++) {
GST_INFO ("starting test-loop %d", i);
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
g_idle_add ((GSourceFunc) set_playing, bin);
GST_INFO ("playing");
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
gst_consistency_checker_reset (consist);
}
/* cleanup */
GST_INFO ("cleaning up");
gst_consistency_checker_free (consist);
if (main_loop)
g_main_loop_unref (main_loop);
if (play_seek_event)
gst_event_unref (play_seek_event);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
/* check if adding pads work as expected */
GST_START_TEST (test_add_pad)
{
GstElement *bin, *src1, *src2, *audiomixer, *sink;
GstBus *bus;
GstPad *srcpad;
gboolean res;
GstStateChangeReturn state_res;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "num-buffers", 4, NULL);
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
/* one buffer less, we connect with 1 buffer of delay */
g_object_set (src2, "num-buffers", 3, NULL);
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL);
res = gst_element_link (src1, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
srcpad = gst_element_get_static_pad (audiomixer, "src");
gst_object_unref (srcpad);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
GST_INFO ("starting test");
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* add other element */
gst_bin_add_many (GST_BIN (bin), src2, NULL);
/* now link the second element */
res = gst_element_link (src2, audiomixer);
fail_unless (res == TRUE, NULL);
/* set to PAUSED as well */
state_res = gst_element_set_state (src2, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* now play all */
g_idle_add ((GSourceFunc) set_playing, bin);
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* cleanup */
g_main_loop_unref (main_loop);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
/* check if removing pads work as expected */
GST_START_TEST (test_remove_pad)
{
GstElement *bin, *src, *audiomixer, *sink;
GstBus *bus;
GstPad *pad, *srcpad;
gboolean res;
GstStateChangeReturn state_res;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src = gst_element_factory_make ("audiotestsrc", "src");
g_object_set (src, "num-buffers", 4, NULL);
g_object_set (src, "wave", 4, NULL);
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL);
res = gst_element_link (src, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
/* create an unconnected sinkpad in audiomixer */
pad = gst_element_get_request_pad (audiomixer, "sink_%u");
fail_if (pad == NULL, NULL);
srcpad = gst_element_get_static_pad (audiomixer, "src");
gst_object_unref (srcpad);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
GST_INFO ("starting test");
/* prepare playing, this will not preroll as audiomixer is waiting
* on the unconnected sinkpad. */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion for one second, will return ASYNC */
state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC);
/* get rid of the pad now, audiomixer should stop waiting on it and
* continue the preroll */
gst_element_release_request_pad (audiomixer, pad);
gst_object_unref (pad);
/* wait for completion, should work now */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* now play all */
g_idle_add ((GSourceFunc) set_playing, bin);
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* cleanup */
g_main_loop_unref (main_loop);
gst_bus_remove_signal_watch (bus);
gst_object_unref (G_OBJECT (bus));
gst_object_unref (G_OBJECT (bin));
}
GST_END_TEST;
static GstBuffer *handoff_buffer = NULL;
static void
handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_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_buffer_replace (&handoff_buffer, buffer);
}
/* check if clipping works as expected */
GST_START_TEST (test_clip)
{
GstSegment segment;
GstElement *bin, *audiomixer, *sink;
GstBus *bus;
GstPad *sinkpad;
gboolean res;
GstStateChangeReturn state_res;
GstFlowReturn ret;
GstEvent *event;
GstBuffer *buffer;
GstCaps *caps;
GstQuery *drain = gst_query_new_drain ();
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
/* just an audiomixer and a fakesink */
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, NULL);
gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
/* set to playing */
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* create an unconnected sinkpad in audiomixer, should also automatically activate
* the pad */
sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
fail_if (sinkpad == NULL, NULL);
gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
gst_pad_set_caps (sinkpad, caps);
gst_caps_unref (caps);
/* send segment to audiomixer */
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = GST_SECOND;
segment.stop = 2 * GST_SECOND;
segment.time = 0;
event = gst_event_new_segment (&segment);
gst_pad_send_event (sinkpad, event);
/* should be clipped and ok */
buffer = gst_buffer_new_and_alloc (44100);
GST_BUFFER_TIMESTAMP (buffer) = 0;
GST_BUFFER_DURATION (buffer) = 250 * 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);
/* The aggregation is done in a dedicated thread, so we can't
* not know when it is actually going to happen, so we use a DRAIN query
* to wait for it to complete.
*/
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer == NULL);
/* should be partially clipped */
buffer = gst_buffer_new_and_alloc (44100);
GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 250 * 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)));
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer != NULL);
gst_buffer_replace (&handoff_buffer, NULL);
/* should not be clipped */
buffer = gst_buffer_new_and_alloc (44100);
GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 250 * 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);
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;
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);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer == NULL);
gst_element_release_request_pad (audiomixer, sinkpad);
gst_object_unref (sinkpad);
gst_element_set_state (bin, GST_STATE_NULL);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
gst_query_unref (drain);
}
GST_END_TEST;
GST_START_TEST (test_duration_is_max)
{
GstElement *bin, *src[3], *audiomixer, *sink;
GstStateChangeReturn state_res;
GstFormat format = GST_FORMAT_TIME;
gboolean res;
gint64 duration;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
/* 3 sources, an audiomixer and a fakesink */
src[0] = gst_element_factory_make ("audiotestsrc", NULL);
src[1] = gst_element_factory_make ("audiotestsrc", NULL);
src[2] = gst_element_factory_make ("audiotestsrc", NULL);
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
NULL);
gst_element_link (src[0], audiomixer);
gst_element_link (src[1], audiomixer);
gst_element_link (src[2], audiomixer);
gst_element_link (audiomixer, sink);
/* irks, duration is reset on basesrc */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
/* set durations on src */
GST_BASE_SRC (src[0])->segment.duration = 1000;
GST_BASE_SRC (src[1])->segment.duration = 3000;
GST_BASE_SRC (src[2])->segment.duration = 2000;
/* set to playing */
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
fail_unless (res, NULL);
ck_assert_int_eq (duration, 3000);
gst_element_set_state (bin, GST_STATE_NULL);
gst_object_unref (bin);
}
GST_END_TEST;
GST_START_TEST (test_duration_unknown_overrides)
{
GstElement *bin, *src[3], *audiomixer, *sink;
GstStateChangeReturn state_res;
GstFormat format = GST_FORMAT_TIME;
gboolean res;
gint64 duration;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
/* 3 sources, an audiomixer and a fakesink */
src[0] = gst_element_factory_make ("audiotestsrc", NULL);
src[1] = gst_element_factory_make ("audiotestsrc", NULL);
src[2] = gst_element_factory_make ("audiotestsrc", NULL);
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
NULL);
gst_element_link (src[0], audiomixer);
gst_element_link (src[1], audiomixer);
gst_element_link (src[2], audiomixer);
gst_element_link (audiomixer, sink);
/* irks, duration is reset on basesrc */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
/* set durations on src */
GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE;
GST_BASE_SRC (src[1])->segment.duration = 3000;
GST_BASE_SRC (src[2])->segment.duration = 2000;
/* set to playing */
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
/* wait for completion */
state_res =
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
GST_CLOCK_TIME_NONE);
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
fail_unless (res, NULL);
ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE);
gst_element_set_state (bin, GST_STATE_NULL);
gst_object_unref (bin);
}
GST_END_TEST;
static gboolean looped = FALSE;
static void
loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin)
{
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
GST_MESSAGE_SRC (message), message);
if (looped) {
g_main_loop_quit (main_loop);
} else {
GstEvent *seek_event;
gboolean res;
seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_SEGMENT,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
res = gst_element_send_event (bin, seek_event);
fail_unless (res == TRUE, NULL);
looped = TRUE;
}
}
GST_START_TEST (test_loop)
{
GstElement *bin, *src1, *src2, *audiomixer, *sink;
GstBus *bus;
GstEvent *seek_event;
GstStateChangeReturn state_res;
gboolean res;
GST_INFO ("preparing test");
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
res = gst_element_link (src1, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (src2, audiomixer);
fail_unless (res == TRUE, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, (GstClockTime) 0,
GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bus, "message::segment-done",
(GCallback) loop_segment_done, bin);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
GST_INFO ("starting test");
/* prepare playing */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for completion */
state_res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
res = gst_element_send_event (bin, seek_event);
fail_unless (res == TRUE, NULL);
/* run pipeline */
g_idle_add ((GSourceFunc) set_playing, bin);
GST_INFO ("running main loop");
g_main_loop_run (main_loop);
state_res = gst_element_set_state (bin, GST_STATE_NULL);
/* cleanup */
g_main_loop_unref (main_loop);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
}
GST_END_TEST;
GST_START_TEST (test_flush_start_flush_stop)
{
GstPadTemplate *sink_template;
GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src;
GstElement *pipeline, *src1, *src2, *audiomixer, *sink;
GST_INFO ("preparing test");
/* build pipeline */
pipeline = gst_pipeline_new ("pipeline");
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "wave", 4, NULL); /* silence */
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "wave", 4, NULL); /* silence */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL);
sink_template =
gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer),
"sink_%u");
fail_unless (GST_IS_PAD_TEMPLATE (sink_template));
sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
srcpad1 = gst_element_get_static_pad (src1, "src");
gst_pad_link (srcpad1, sinkpad1);
sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
tmppad = gst_element_get_static_pad (src2, "src");
gst_pad_link (tmppad, sinkpad2);
gst_object_unref (tmppad);
gst_element_link (audiomixer, sink);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
fail_unless (gst_element_get_state (pipeline, NULL, NULL,
GST_CLOCK_TIME_NONE) == GST_STATE_CHANGE_SUCCESS);
audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
/* Hold the streamlock to make sure the flush stop is not between
the attempted push of a segment event and of the following buffer. */
GST_PAD_STREAM_LOCK (srcpad1);
gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
GST_PAD_STREAM_UNLOCK (srcpad1);
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
gst_object_unref (audiomixer_src);
gst_element_release_request_pad (audiomixer, sinkpad1);
gst_object_unref (sinkpad1);
gst_element_release_request_pad (audiomixer, sinkpad2);
gst_object_unref (sinkpad2);
gst_object_unref (srcpad1);
/* cleanup */
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
static void
handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
GstPad * pad, gpointer user_data)
{
GList **received_buffers = user_data;
GST_DEBUG ("got buffer %p", buffer);
*received_buffers =
g_list_append (*received_buffers, gst_buffer_ref (buffer));
}
typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2);
typedef void (*CheckBuffersFunction) (GList * buffers);
static void
run_sync_test (SendBuffersFunction send_buffers,
CheckBuffersFunction check_buffers)
{
GstSegment segment;
GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
GstBus *bus;
GstPad *sinkpad1, *sinkpad2;
GstPad *queue1_sinkpad, *queue2_sinkpad;
GstPad *pad;
gboolean res;
GstStateChangeReturn state_res;
GstEvent *event;
GstCaps *caps;
GList *received_buffers = NULL;
GST_INFO ("preparing test");
main_loop = g_main_loop_new (NULL, FALSE);
/* build pipeline */
bin = gst_pipeline_new ("pipeline");
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
/* just an audiomixer and a fakesink */
queue1 = gst_element_factory_make ("queue", "queue1");
queue2 = gst_element_factory_make ("queue", "queue2");
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
sink = gst_element_factory_make ("fakesink", "sink");
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
&received_buffers);
gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
res = gst_element_link (audiomixer, sink);
fail_unless (res == TRUE, NULL);
/* set to paused */
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* create an unconnected sinkpad in audiomixer, should also automatically activate
* the pad */
sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
fail_if (sinkpad1 == NULL, NULL);
queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
pad = gst_element_get_static_pad (queue1, "src");
fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
gst_object_unref (pad);
sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
fail_if (sinkpad2 == NULL, NULL);
queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
pad = gst_element_get_static_pad (queue2, "src");
fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
gst_object_unref (pad);
gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
gst_pad_set_caps (queue1_sinkpad, caps);
gst_pad_set_caps (queue2_sinkpad, caps);
gst_caps_unref (caps);
/* send segment to audiomixer */
gst_segment_init (&segment, GST_FORMAT_TIME);
event = gst_event_new_segment (&segment);
gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
gst_pad_send_event (queue2_sinkpad, event);
/* Push buffers */
send_buffers (queue1_sinkpad, queue2_sinkpad);
/* Set PLAYING */
g_idle_add ((GSourceFunc) set_playing, bin);
/* Collect buffers and messages */
g_main_loop_run (main_loop);
/* Here we get once we got EOS, for errors we failed */
check_buffers (received_buffers);
g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
gst_element_release_request_pad (audiomixer, sinkpad1);
gst_object_unref (sinkpad1);
gst_object_unref (queue1_sinkpad);
gst_element_release_request_pad (audiomixer, sinkpad2);
gst_object_unref (sinkpad2);
gst_object_unref (queue2_sinkpad);
gst_element_set_state (bin, GST_STATE_NULL);
gst_bus_remove_signal_watch (bus);
gst_object_unref (bus);
gst_object_unref (bin);
g_main_loop_unref (main_loop);
}
static void
send_buffers_sync (GstPad * pad1, GstPad * pad2)
{
GstBuffer *buffer;
GstMapInfo map;
GstFlowReturn ret;
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad2, gst_event_new_eos ());
}
static void
check_buffers_sync (GList * received_buffers)
{
GstBuffer *buffer;
GList *l;
gint i;
GstMapInfo map;
/* Should have 8 * 0.5s buffers */
fail_unless_equals_int (g_list_length (received_buffers), 8);
for (i = 0, l = received_buffers; l; l = l->next, i++) {
buffer = l->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[map.size - 1] == 0);
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[map.size - 1] == 0);
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
fail_unless (map.data[0] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
fail_unless (map.data[0] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (buffer, &map);
}
}
GST_START_TEST (test_sync)
{
run_sync_test (send_buffers_sync, check_buffers_sync);
}
GST_END_TEST;
static void
send_buffers_sync_discont (GstPad * pad1, GstPad * pad2)
{
GstBuffer *buffer;
GstMapInfo map;
GstFlowReturn ret;
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad2, gst_event_new_eos ());
}
static void
check_buffers_sync_discont (GList * received_buffers)
{
GstBuffer *buffer;
GList *l;
gint i;
GstMapInfo map;
/* Should have 8 * 0.5s buffers */
fail_unless_equals_int (g_list_length (received_buffers), 8);
for (i = 0, l = received_buffers; l; l = l->next, i++) {
buffer = l->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[map.size - 1] == 0);
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[map.size - 1] == 0);
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
fail_unless (map.data[0] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
fail_unless (map.data[0] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (buffer, &map);
}
}
GST_START_TEST (test_sync_discont)
{
run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont);
}
GST_END_TEST;
static void
send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2)
{
GstBuffer *buffer;
GstMapInfo map;
GstFlowReturn ret;
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 750 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 1, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 1750 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 1750 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = gst_buffer_new_and_alloc (2000);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 2, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = 2750 * GST_MSECOND;
GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
GST_DEBUG ("pushing buffer %p", buffer);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad2, gst_event_new_eos ());
}
static void
check_buffers_sync_unaligned (GList * received_buffers)
{
GstBuffer *buffer;
GList *l;
gint i;
GstMapInfo map;
/* Should have 8 * 0.5s buffers */
fail_unless_equals_int (g_list_length (received_buffers), 8);
for (i = 0, l = received_buffers; l; l = l->next, i++) {
buffer = l->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[map.size - 1] == 0);
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
fail_unless (map.data[0] == 0);
fail_unless (map.data[499] == 0);
fail_unless (map.data[500] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[map.size - 1] == 1);
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
fail_unless (map.data[0] == 1);
fail_unless (map.data[499] == 1);
fail_unless (map.data[500] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[499] == 3);
fail_unless (map.data[500] == 3);
fail_unless (map.data[map.size - 1] == 3);
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
fail_unless (map.data[0] == 3);
fail_unless (map.data[499] == 3);
fail_unless (map.data[500] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
fail_unless (map.data[0] == 2);
fail_unless (map.data[499] == 2);
fail_unless (map.data[500] == 2);
fail_unless (map.data[map.size - 1] == 2);
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
fail_unless (map.size == 500);
fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND);
fail_unless (map.data[0] == 2);
fail_unless (map.data[499] == 2);
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (buffer, &map);
}
}
GST_START_TEST (test_sync_unaligned)
{
run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned);
}
GST_END_TEST;
GST_START_TEST (test_segment_base_handling)
{
GstElement *pipeline, *sink, *mix, *src1, *src2;
GstPad *srcpad, *sinkpad;
GstClockTime end_time;
GstSample *last_sample = NULL;
GstSample *sample;
GstBuffer *buf;
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
"channels", G_TYPE_INT, 2, NULL);
pipeline = gst_pipeline_new ("pipeline");
mix = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("appsink", "sink");
g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
gst_caps_unref (caps);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL);
fail_unless (gst_element_link (mix, sink));
srcpad = gst_element_get_static_pad (src1, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_1");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
srcpad = gst_element_get_static_pad (src2, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_2");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
do {
g_signal_emit_by_name (sink, "pull-sample", &sample);
if (sample == NULL)
break;
if (last_sample)
gst_sample_unref (last_sample);
last_sample = sample;
} while (TRUE);
buf = gst_sample_get_buffer (last_sample);
end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
gst_sample_unref (last_sample);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
static void
set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
GstClockTime end, gdouble end_value)
{
GstControlSource *cs;
GstTimedValueControlSource *tvcs;
cs = gst_interpolation_control_source_new ();
fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
"volume", cs)));
/* set volume interpolation mode */
g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
tvcs = (GstTimedValueControlSource *) cs;
fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
gst_object_unref (cs);
}
GST_START_TEST (test_sinkpad_property_controller)
{
GstBus *bus;
GstMessage *msg;
GstElement *pipeline, *sink, *mix, *src1;
GstPad *srcpad, *sinkpad;
GError *error = NULL;
gchar *debug;
pipeline = gst_pipeline_new ("pipeline");
mix = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "num-buffers", 100, NULL);
gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
fail_unless (gst_element_link (mix, sink));
srcpad = gst_element_get_static_pad (src1, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_0");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &error, &debug);
g_printerr ("ERROR from element %s: %s\n",
GST_OBJECT_NAME (msg->src), error->message);
g_printerr ("Debug info: %s\n", debug);
g_error_free (error);
g_free (debug);
break;
case GST_MESSAGE_EOS:
break;
default:
g_assert_not_reached ();
}
gst_message_unref (msg);
g_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
static Suite *
audiomixer_suite (void)
{
Suite *s = suite_create ("audiomixer");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_caps);
tcase_add_test (tc_chain, test_filter_caps);
tcase_add_test (tc_chain, test_event);
tcase_add_test (tc_chain, test_play_twice);
tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again);
tcase_add_test (tc_chain, test_live_seeking);
tcase_add_test (tc_chain, test_add_pad);
tcase_add_test (tc_chain, test_remove_pad);
tcase_add_test (tc_chain, test_clip);
tcase_add_test (tc_chain, test_duration_is_max);
tcase_add_test (tc_chain, test_duration_unknown_overrides);
tcase_add_test (tc_chain, test_loop);
tcase_add_test (tc_chain, test_flush_start_flush_stop);
tcase_add_test (tc_chain, test_sync);
tcase_add_test (tc_chain, test_sync_discont);
tcase_add_test (tc_chain, test_sync_unaligned);
tcase_add_test (tc_chain, test_segment_base_handling);
tcase_add_test (tc_chain, test_sinkpad_property_controller);
/* Use a longer timeout */
#ifdef HAVE_VALGRIND
if (RUNNING_ON_VALGRIND) {
tcase_set_timeout (tc_chain, 5 * 60);
} else
#endif
{
/* this is shorter than the default 60 seconds?! (tpm) */
/* tcase_set_timeout (tc_chain, 6); */
}
return s;
}
GST_CHECK_MAIN (audiomixer);
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