--- /dev/null
+/* GStreamer
+ *
+ * unit test for adder
+ *
+ * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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>
+
+static GMainLoop *main_loop;
+
+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, *adder, *sink;
+ GstBus *bus;
+ GstEvent *seek_event;
+ gboolean 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);
+
+ /* FIXME, fakesrc with default setting will produce 0 sized
+ * buffers and incompatible caps for adder that will make
+ * adder EOS and error out */
+ 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 */
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
+
+ res = gst_element_link (src1, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (src2, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ srcpad = gst_element_get_static_pad (adder, "src");
+ consist = gst_consistency_checker_new (srcpad);
+ 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 */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion */
+ res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ res = gst_element_send_event (bin, seek_event);
+ fail_unless (res == TRUE, NULL);
+
+ /* run pipeline */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ fail_unless (position == 2 * GST_SECOND, NULL);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+ gst_consistency_checker_free (consist);
+ gst_object_unref (G_OBJECT (bus));
+ gst_object_unref (G_OBJECT (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;
+
+ 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) {
+ res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_READY);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* prepare playing again */
+ res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ res = gst_element_send_event (GST_ELEMENT (bin),
+ gst_event_ref (play_seek_event));
+ fail_unless (res == TRUE, NULL);
+
+ res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+ } else {
+ g_main_loop_quit (main_loop);
+ }
+ break;
+ default:
+ g_assert_not_reached ();
+ break;
+ }
+}
+
+
+GST_START_TEST (test_play_twice)
+{
+ GstElement *bin, *src1, *src2, *adder, *sink;
+ GstBus *bus;
+ gboolean 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 */
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
+
+ res = gst_element_link (src1, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (src2, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ srcpad = gst_element_get_static_pad (adder, "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 */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+ fail_unless (res == TRUE, NULL);
+
+ GST_INFO ("seeked");
+
+ /* run pipeline */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ fail_unless (play_count == 2, NULL);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+ gst_consistency_checker_free (consist);
+ gst_event_ref (play_seek_event);
+ gst_object_unref (G_OBJECT (bus));
+ gst_object_unref (G_OBJECT (bin));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_twice_then_add_and_play_again)
+{
+ GstElement *bin, *src1, *src2, *src3, *adder, *sink;
+ GstBus *bus;
+ gboolean 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 */
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
+
+ srcpad = gst_element_get_static_pad (adder, "src");
+ consist = gst_consistency_checker_new (srcpad);
+ gst_object_unref (srcpad);
+
+ res = gst_element_link (src1, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (src2, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, 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 */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+ fail_unless (res == TRUE, NULL);
+
+ GST_INFO ("seeked");
+
+ /* run pipeline */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_READY);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ fail_unless (play_count == 2, NULL);
+
+ /* 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, adder);
+ fail_unless (res == TRUE, NULL);
+ }
+
+ gst_consistency_checker_reset (consist);
+ }
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+ gst_event_ref (play_seek_event);
+ gst_consistency_checker_free (consist);
+ gst_object_unref (G_OBJECT (bus));
+ gst_object_unref (G_OBJECT (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;
+ }
+}
+
+
+/* test failing seeks on live-sources */
+GST_START_TEST (test_live_seeking)
+{
+ GstElement *bin, *src1, *src2, *ac1, *ac2, *adder, *sink;
+ GstBus *bus;
+ gboolean res;
+ GstPad *srcpad;
+ gint i;
+ GstStreamConsistency *consist;
+
+ 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);
+
+ /* normal audiosources behave differently than audiotestsrc */
+#if 0
+ src1 = gst_element_factory_make ("audiotestsrc", "src1");
+ g_object_set (src1, "wave", 4, "is-live", TRUE, NULL); /* silence */
+#else
+ src1 = gst_element_factory_make ("alsasrc", "src1");
+ if (!src1) {
+ GST_INFO ("no audiosrc, skipping");
+ goto cleanup;
+ }
+ /* Test that the audio source can get to paused, else skip */
+ res = gst_element_set_state (src1, GST_STATE_PAUSED);
+ (void) gst_element_set_state (src1, GST_STATE_NULL);
+ gst_object_unref (src1);
+
+ if (res == GST_STATE_CHANGE_FAILURE)
+ goto cleanup;
+ src1 = gst_element_factory_make ("alsasrc", "src1");
+
+ /* live sources ignore seeks, force eos after 2 sec (4 buffers half second
+ * each) - don't use autoaudiosrc, as then we can't set anything here */
+ g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL);
+#endif
+ ac1 = gst_element_factory_make ("audioconvert", "ac1");
+ src2 = gst_element_factory_make ("audiotestsrc", "src2");
+ g_object_set (src2, "wave", 4, NULL); /* silence */
+ ac2 = gst_element_factory_make ("audioconvert", "ac2");
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src1, ac1, src2, ac2, adder, sink, NULL);
+
+ res = gst_element_link (src1, ac1);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (ac1, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (src2, ac2);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (ac2, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ 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 (adder, "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 */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE);
+
+ /* wait for completion */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
+#if 1
+ fail_unless (res == TRUE, NULL);
+#else
+ /* adder is picky, if a single seek fails it totaly fails */
+ fail_unless (res == FALSE, NULL);
+#endif
+
+ GST_INFO ("seeked");
+
+ /* run pipeline */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ GST_INFO ("playing");
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ gst_consistency_checker_reset (consist);
+ }
+
+ /* cleanup */
+cleanup:
+ GST_INFO ("cleaning up");
+ if (main_loop)
+ g_main_loop_unref (main_loop);
+ if (play_seek_event)
+ gst_event_unref (play_seek_event);
+ gst_object_unref (G_OBJECT (bus));
+ gst_object_unref (G_OBJECT (bin));
+}
+
+GST_END_TEST;
+
+/* check if adding pads work as expected */
+GST_START_TEST (test_add_pad)
+{
+ GstElement *bin, *src1, *src2, *adder, *sink;
+ GstBus *bus;
+ GstPad *srcpad;
+ GstStreamConsistency *consist;
+ 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, "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 */
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src1, adder, sink, NULL);
+
+ res = gst_element_link (src1, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ srcpad = gst_element_get_static_pad (adder, "src");
+ consist = gst_consistency_checker_new (srcpad);
+ 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 */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* add other element */
+ gst_bin_add_many (GST_BIN (bin), src2, NULL);
+
+ /* now link the second element */
+ res = gst_element_link (src2, adder);
+ fail_unless (res == TRUE, NULL);
+
+ /* set to PAUSED as well */
+ res = gst_element_set_state (src2, GST_STATE_PAUSED);
+
+ /* now play all */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+ gst_object_unref (G_OBJECT (bus));
+ gst_object_unref (G_OBJECT (bin));
+}
+
+GST_END_TEST;
+
+/* check if removing pads work as expected */
+GST_START_TEST (test_remove_pad)
+{
+ GstElement *bin, *src, *adder, *sink;
+ GstBus *bus;
+ GstPad *pad, *srcpad;
+ gboolean res;
+ 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);
+
+ src = gst_element_factory_make ("audiotestsrc", "src");
+ g_object_set (src, "num-buffers", 4, NULL);
+ g_object_set (src, "wave", 4, NULL);
+ adder = gst_element_factory_make ("adder", "adder");
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src, adder, sink, NULL);
+
+ res = gst_element_link (src, adder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ /* create an unconnected sinkpad in adder */
+ pad = gst_element_get_request_pad (adder, "sink%d");
+ fail_if (pad == NULL, NULL);
+
+ srcpad = gst_element_get_static_pad (adder, "src");
+ consist = gst_consistency_checker_new (srcpad);
+ 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 adder is waiting
+ * on the unconnected sinkpad. */
+ res = gst_element_set_state (bin, GST_STATE_PAUSED);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* wait for completion for one second, will return ASYNC */
+ res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
+ fail_unless (res == GST_STATE_CHANGE_ASYNC, NULL);
+
+ /* get rid of the pad now, adder should stop waiting on it and
+ * continue the preroll */
+ gst_element_release_request_pad (adder, pad);
+ gst_object_unref (pad);
+
+ /* wait for completion, should work now */
+ res =
+ gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
+ GST_CLOCK_TIME_NONE);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* now play all */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ g_main_loop_run (main_loop);
+
+ res = gst_element_set_state (bin, GST_STATE_NULL);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+ 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 %p", buffer);
+ gst_buffer_replace (&handoff_buffer, buffer);
+}
+
+/* check if clipping works as expected */
+GST_START_TEST (test_clip)
+{
+ GstElement *bin, *adder, *sink;
+ GstBus *bus;
+ GstPad *sinkpad;
+ gboolean res;
+ GstFlowReturn ret;
+ GstEvent *event;
+ GstBuffer *buffer;
+ GstCaps *caps;
+
+ 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 adder and a fakesink */
+ adder = gst_element_factory_make ("adder", "adder");
+ 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), adder, sink, NULL);
+
+ res = gst_element_link (adder, sink);
+ fail_unless (res == TRUE, NULL);
+
+ /* set to playing */
+ res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
+
+ /* create an unconnected sinkpad in adder, should also automatically activate
+ * the pad */
+ sinkpad = gst_element_get_request_pad (adder, "sink%d");
+ fail_if (sinkpad == NULL, NULL);
+
+ /* send segment to adder */
+ event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
+ GST_SECOND, 2 * GST_SECOND, 0);
+ gst_pad_send_event (sinkpad, event);
+
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, 44100,
+ "channels", G_TYPE_INT, 2,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "width", G_TYPE_INT, 16,
+ "depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+ /* 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_buffer_set_caps (buffer, caps);
+ GST_DEBUG ("pushing buffer %p", buffer);
+ ret = gst_pad_chain (sinkpad, buffer);
+ fail_unless (ret == GST_FLOW_OK);
+ 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_buffer_set_caps (buffer, caps);
+ GST_DEBUG ("pushing buffer %p", buffer);
+ ret = gst_pad_chain (sinkpad, buffer);
+ fail_unless (ret == GST_FLOW_OK);
+ 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_buffer_set_caps (buffer, caps);
+ GST_DEBUG ("pushing buffer %p", buffer);
+ ret = gst_pad_chain (sinkpad, buffer);
+ fail_unless (ret == GST_FLOW_OK);
+ fail_unless (handoff_buffer != NULL);
+ gst_buffer_replace (&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_buffer_set_caps (buffer, caps);
+ GST_DEBUG ("pushing buffer %p", buffer);
+ ret = gst_pad_chain (sinkpad, buffer);
+ fail_unless (ret == GST_FLOW_OK);
+ fail_unless (handoff_buffer == NULL);
+
+
+}
+
+GST_END_TEST;
+
+static Suite *
+adder_suite (void)
+{
+ Suite *s = suite_create ("adder");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ 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);
+
+ /* 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 (adder);
projects / mafwsubrenderer / blobdiff