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gstreamer/tests/check/elements/audiomixer.c
Tim-Philipp Müller 6f6c73b223 tests: audiomixer: fix leaks in unit test
2019-03-06 10:51:40 +00:00

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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 <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;
/* fixtures */
static void
test_setup (void)
{
main_loop = g_main_loop_new (NULL, FALSE);
}
static void
test_teardown (void)
{
g_main_loop_unref (main_loop);
main_loop = NULL;
}
/* some test helpers */
static GstElement *
setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter)
{
GstElement *pipeline, *src, *sink;
gint i;
pipeline = gst_pipeline_new ("pipeline");
if (!audiomixer) {
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
}
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL);
if (capsfilter) {
gst_bin_add (GST_BIN (pipeline), capsfilter);
gst_element_link_many (audiomixer, capsfilter, sink, NULL);
} else {
gst_element_link (audiomixer, sink);
}
for (i = 0; i < num_srcs; i++) {
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (src, "wave", 4, NULL); /* silence */
gst_bin_add (GST_BIN (pipeline), src);
gst_element_link (src, audiomixer);
}
return pipeline;
}
static GstCaps *
get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name)
{
GstElement *sink;
GstCaps *caps;
GstPad *pad;
sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
pad = gst_element_get_static_pad (sink, "sink");
caps = gst_pad_get_current_caps (pad);
gst_object_unref (pad);
gst_object_unref (sink);
return caps;
}
static void
set_state_and_wait (GstElement * pipeline, GstState state)
{
GstStateChangeReturn state_res;
/* prepare paused/playing */
state_res = gst_element_set_state (pipeline, state);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* wait for preroll */
state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
}
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
play_and_wait (GstElement * pipeline)
{
GstStateChangeReturn state_res;
g_idle_add ((GSourceFunc) set_playing, pipeline);
GST_INFO ("running main loop");
g_main_loop_run (main_loop);
state_res = gst_element_set_state (pipeline, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
}
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 GstBuffer *
new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur,
GstBufferFlags flags)
{
GstMapInfo map;
GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes);
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, data, map.size);
gst_buffer_unmap (buffer, &map);
GST_BUFFER_TIMESTAMP (buffer) = ts;
GST_BUFFER_DURATION (buffer) = dur;
if (flags)
GST_BUFFER_FLAG_SET (buffer, flags);
GST_DEBUG ("created buffer %p", buffer);
return buffer;
}
/* make sure downstream gets a CAPS event before buffers are sent */
GST_START_TEST (test_caps)
{
GstElement *pipeline;
GstCaps *caps;
/* build pipeline */
pipeline = setup_pipeline (NULL, 1, NULL);
/* prepare playing */
set_state_and_wait (pipeline, GST_STATE_PAUSED);
/* check caps on fakesink */
caps = get_element_sink_pad_caps (pipeline, "sink");
fail_unless (caps != NULL);
gst_caps_unref (caps);
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, *audiomixer, *capsfilter;
GstCaps *filter_caps, *caps;
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);
capsfilter = gst_element_factory_make ("capsfilter", NULL);
/* build pipeline */
audiomixer = gst_element_factory_make ("audiomixer", NULL);
g_object_set (capsfilter, "caps", filter_caps, NULL);
pipeline = setup_pipeline (audiomixer, 1, capsfilter);
/* prepare playing */
set_state_and_wait (pipeline, GST_STATE_PAUSED);
/* check caps on fakesink */
caps = get_element_sink_pad_caps (pipeline, "sink");
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_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
gst_caps_unref (filter_caps);
}
GST_END_TEST;
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;
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;
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, seek_event);
fail_unless (res == TRUE, NULL);
/* run pipeline */
play_and_wait (bin);
ck_assert_int_eq (position, 2 * GST_SECOND);
/* cleanup */
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,
GstElement * 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 (bin, GST_STATE_READY);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
/* prepare playing again */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
state_res = gst_element_set_state (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, *audiomixer;
GstBus *bus;
gboolean res;
GstPad *srcpad;
GstStreamConsistency *consist;
GST_INFO ("preparing test");
/* build pipeline */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
bin = setup_pipeline (audiomixer, 2, NULL);
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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;
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
play_and_wait (bin);
ck_assert_int_eq (play_count, 2);
/* cleanup */
gst_consistency_checker_free (consist);
gst_event_unref (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, *src, *audiomixer;
GstBus *bus;
gboolean res;
GstStateChangeReturn state_res;
gint i;
GstPad *srcpad;
GstStreamConsistency *consist;
GST_INFO ("preparing test");
/* build pipeline */
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
bin = setup_pipeline (audiomixer, 2, NULL);
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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);
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
play_and_wait (bin);
ck_assert_int_eq (play_count, 2);
/* plug another source */
if (i == 0) {
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (src, "wave", 4, NULL); /* silence */
gst_bin_add (GST_BIN (bin), src);
res = gst_element_link (src, 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 */
gst_event_unref (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 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;
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");
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 */
GST_WARNING ("no real audiosrc found, using audiotestsrc is-live");
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_many (src1, cf, audiomixer, sink, NULL);
fail_unless (res == TRUE, NULL);
/* get the caps for the livesrc, we'll reuse this for the non-live source */
set_state_and_wait (bin, GST_STATE_PLAYING);
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);
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);
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
fail_unless (res == TRUE, NULL);
GST_INFO ("seeked");
/* run pipeline */
play_and_wait (bin);
gst_consistency_checker_reset (consist);
}
/* cleanup */
GST_INFO ("cleaning up");
gst_consistency_checker_free (consist);
if (play_seek_event)
gst_event_unref (play_seek_event);
gst_bus_remove_signal_watch (bus);
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, "wave", /* silence */ 4, NULL);
src2 = gst_element_factory_make ("audiotestsrc", "src2");
/* one buffer less, we connect with 1 buffer of delay */
g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL);
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);
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
/* 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 */
play_and_wait (bin);
/* cleanup */
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, "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);
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 */
play_and_wait (bin);
/* cleanup */
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 PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT,
gst_buffer_get_size (buffer), buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (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");
g_object_set (audiomixer, "output-buffer-duration", 50 * 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_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 = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0);
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
* 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 = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND,
GST_BUFFER_FLAG_DISCONT);
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);
ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND);
gst_buffer_replace (&handoff_buffer, NULL);
/* should not be clipped */
buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0);
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);
ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND);
gst_buffer_replace (&handoff_buffer, NULL);
fail_unless (handoff_buffer == NULL);
/* should be clipped and ok */
buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND,
GST_BUFFER_FLAG_DISCONT);
GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT
" END is %" GST_TIME_FORMAT,
buffer,
GST_TIME_ARGS (GST_BUFFER_PTS (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 */
set_state_and_wait (bin, GST_STATE_PLAYING);
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 */
set_state_and_wait (bin, GST_STATE_PLAYING);
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;
GstBus *bus;
GstEvent *seek_event;
gboolean res;
GST_INFO ("preparing test");
/* build pipeline */
bin = setup_pipeline (NULL, 2, NULL);
bus = gst_element_get_bus (bin);
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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);
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 */
set_state_and_wait (bin, GST_STATE_PAUSED);
res = gst_element_send_event (bin, seek_event);
fail_unless (res == TRUE, NULL);
/* run pipeline */
play_and_wait (bin);
fail_unless (looped);
/* cleanup */
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);
/* prepare playing */
set_state_and_wait (pipeline, GST_STATE_PLAYING);
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");
/* 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);
}
static void
send_buffers_sync (GstPad * pad1, GstPad * pad2)
{
GstBuffer *buffer;
GstFlowReturn ret;
buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
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;
GstFlowReturn ret;
buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND,
GST_BUFFER_FLAG_DISCONT);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
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;
GstFlowReturn ret;
buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad1, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_send_event (pad1, gst_event_new_eos ());
buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
ret = gst_pad_chain (pad2, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0);
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);
/* 50 buffers of 1/10 sec = 5 sec */
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);
/* set a pad offset of another 5 seconds */
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 void
change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
GstElement * capsfilter)
{
GstCaps *caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
g_object_set (capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL);
g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter);
}
/* In this test, we create an input buffer with a duration of 2 seconds,
* and require the audiomixer to output 1 second long buffers.
* The input buffer will thus be mixed twice, and the audiomixer will
* output two buffers.
*
* After audiomixer has output a first buffer, we change its output format
* from S8 to S32. As our sample rate stays the same at 10 fps, and we use
* mono, the first buffer should be 10 bytes long, and the second 40.
*
* The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes.
* We verify that the second buffer contains 5 0-valued integers, and
* 5 1 << 24 valued integers.
*/
GST_START_TEST (test_change_output_caps)
{
GstSegment segment;
GstElement *bin, *audiomixer, *capsfilter, *sink;
GstBus *bus;
GstPad *sinkpad;
gboolean res;
GstStateChangeReturn state_res;
GstFlowReturn ret;
GstEvent *event;
GstBuffer *buffer;
GstCaps *caps;
GstQuery *drain = gst_query_new_drain ();
GstMapInfo inmap;
GstMapInfo outmap;
gsize i;
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);
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL);
capsfilter = gst_element_factory_make ("capsfilter", NULL);
sink = gst_element_factory_make ("fakesink", "sink");
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter);
gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
fail_unless (res == TRUE, NULL);
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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, "S8",
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
gst_pad_set_caps (sinkpad, caps);
g_object_set (capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = 0;
segment.stop = 2 * GST_SECOND;
segment.time = 0;
event = gst_event_new_segment (&segment);
gst_pad_send_event (sinkpad, event);
gst_buffer_replace (&handoff_buffer, NULL);
buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0);
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
memset (inmap.data + 15, 1, 5);
gst_buffer_unmap (buffer, &inmap);
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
gst_pad_query (sinkpad, drain);
fail_unless (handoff_buffer != NULL);
fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40);
gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
for (i = 0; i < 10; i++) {
guint32 sample;
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
#else
sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
#endif
if (i < 5) {
fail_unless_equals_int (sample, 0);
} else {
fail_unless_equals_int (sample, 1 << 24);
}
}
gst_buffer_unmap (handoff_buffer, &outmap);
gst_clear_buffer (&handoff_buffer);
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;
/* In this test, we create two input buffers with a duration of 1 second,
* and require the audiomixer to output 1.5 second long buffers.
*
* After we have input two buffers, we change the output format
* from S8 to S32, then push a last buffer.
*
* This makes audioaggregator convert its "half-mixed" current_buffer,
* we can then ensure that the second output buffer is as expected.
*/
GST_START_TEST (test_change_output_caps_mid_output_buffer)
{
GstSegment segment;
GstElement *bin, *audiomixer, *capsfilter, *sink;
GstBus *bus;
GstPad *sinkpad;
gboolean res;
GstStateChangeReturn state_res;
GstFlowReturn ret;
GstEvent *event;
GstBuffer *buffer;
GstCaps *caps;
GstQuery *drain;
GstMapInfo inmap;
GstMapInfo outmap;
guint i;
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);
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
g_object_set (audiomixer, "output-buffer-duration", 1500 * GST_MSECOND, NULL);
capsfilter = gst_element_factory_make ("capsfilter", NULL);
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
fail_unless (res == TRUE, NULL);
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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, "S8",
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
gst_pad_set_caps (sinkpad, caps);
g_object_set (capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = 0;
segment.stop = 3 * GST_SECOND;
segment.time = 0;
event = gst_event_new_segment (&segment);
gst_pad_send_event (sinkpad, event);
buffer = new_buffer (10, 0, 0, 1 * GST_SECOND, 0);
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
buffer = new_buffer (10, 0, 1 * GST_SECOND, 1 * GST_SECOND, 0);
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
memset (inmap.data, 1, 10);
gst_buffer_unmap (buffer, &inmap);
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
drain = gst_query_new_drain ();
gst_pad_query (sinkpad, drain);
gst_query_unref (drain);
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
g_object_set (capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
gst_buffer_replace (&handoff_buffer, NULL);
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
buffer = new_buffer (10, 0, 2 * GST_SECOND, 1 * GST_SECOND, 0);
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
memset (inmap.data, 0, 10);
gst_buffer_unmap (buffer, &inmap);
ret = gst_pad_chain (sinkpad, buffer);
ck_assert_int_eq (ret, GST_FLOW_OK);
drain = gst_query_new_drain ();
gst_pad_query (sinkpad, drain);
gst_query_unref (drain);
fail_unless (handoff_buffer);
fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 60);
gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
for (i = 0; i < 15; i++) {
guint32 sample;
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
#else
sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
#endif
if (i < 5) {
fail_unless_equals_int (sample, 1 << 24);
} else {
fail_unless_equals_int (sample, 0);
}
}
gst_buffer_unmap (handoff_buffer, &outmap);
gst_clear_buffer (&handoff_buffer);
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_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);
tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
tcase_add_test (tc_chain, test_change_output_caps);
tcase_add_test (tc_chain, test_change_output_caps_mid_output_buffer);
/* 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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