mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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0acef1dbe9
... such as setting input caps, ensuring to unmap and bearing in mind that gst_buffer_make_writable usually does not provide a separate memory copy/area.
889 lines
27 KiB
C
889 lines
27 KiB
C
/* GStreamer
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*
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* unit test for adder
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#endif
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstconsistencychecker.h>
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static GMainLoop *main_loop;
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static void
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message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_EOS:
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g_main_loop_quit (main_loop);
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break;
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case GST_MESSAGE_WARNING:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_warning (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_ERROR:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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g_main_loop_quit (main_loop);
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break;
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}
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default:
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break;
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}
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}
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static GstFormat format = GST_FORMAT_UNDEFINED;
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static gint64 position = -1;
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static void
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test_event_message_received (GstBus * bus, GstMessage * message,
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GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_SEGMENT_DONE:
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gst_message_parse_segment_done (message, &format, &position);
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GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
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g_main_loop_quit (main_loop);
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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GST_START_TEST (test_event)
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{
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GstElement *bin, *src1, *src2, *adder, *sink;
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GstBus *bus;
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GstEvent *seek_event;
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gboolean res;
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GstPad *srcpad, *sinkpad;
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GstStreamConsistency *chk_1, *chk_2, *chk_3;
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GST_INFO ("preparing test");
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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src1 = gst_element_factory_make ("audiotestsrc", "src1");
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g_object_set (src1, "wave", 4, NULL); /* silence */
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src2 = gst_element_factory_make ("audiotestsrc", "src2");
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g_object_set (src2, "wave", 4, NULL); /* silence */
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adder = gst_element_factory_make ("adder", "adder");
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
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res = gst_element_link (src1, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (src2, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (adder, sink);
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fail_unless (res == TRUE, NULL);
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srcpad = gst_element_get_static_pad (adder, "src");
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chk_3 = gst_consistency_checker_new (srcpad);
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gst_object_unref (srcpad);
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/* create consistency checkers for the pads */
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srcpad = gst_element_get_static_pad (src1, "src");
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chk_1 = gst_consistency_checker_new (srcpad);
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sinkpad = gst_pad_get_peer (srcpad);
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gst_consistency_checker_add_pad (chk_3, sinkpad);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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srcpad = gst_element_get_static_pad (src2, "src");
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chk_2 = gst_consistency_checker_new (srcpad);
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sinkpad = gst_pad_get_peer (srcpad);
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gst_consistency_checker_add_pad (chk_3, sinkpad);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
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GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
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GST_SEEK_TYPE_SET, (GstClockTime) 0,
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GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
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format = GST_FORMAT_UNDEFINED;
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position = -1;
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main_loop = g_main_loop_new (NULL, FALSE);
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g_signal_connect (bus, "message::segment-done",
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(GCallback) test_event_message_received, bin);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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GST_INFO ("starting test");
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/* prepare playing */
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res = gst_element_set_state (bin, GST_STATE_PAUSED);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* wait for completion */
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res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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res = gst_element_send_event (bin, seek_event);
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fail_unless (res == TRUE, NULL);
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/* run pipeline */
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res = gst_element_set_state (bin, GST_STATE_PLAYING);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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g_main_loop_run (main_loop);
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res = gst_element_set_state (bin, GST_STATE_NULL);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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fail_unless (position == 2 * GST_SECOND, NULL);
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/* cleanup */
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g_main_loop_unref (main_loop);
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gst_consistency_checker_free (chk_1);
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gst_consistency_checker_free (chk_2);
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gst_consistency_checker_free (chk_3);
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gst_object_unref (G_OBJECT (bus));
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gst_object_unref (G_OBJECT (bin));
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}
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GST_END_TEST;
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static guint play_count = 0;
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static GstEvent *play_seek_event = NULL;
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static void
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test_play_twice_message_received (GstBus * bus, GstMessage * message,
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GstPipeline * bin)
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{
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gboolean res;
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_SEGMENT_DONE:
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play_count++;
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if (play_count == 1) {
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res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_READY);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* prepare playing again */
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res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PAUSED);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* wait for completion */
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res =
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gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
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GST_CLOCK_TIME_NONE);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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res = gst_element_send_event (GST_ELEMENT (bin),
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gst_event_ref (play_seek_event));
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fail_unless (res == TRUE, NULL);
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res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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} else {
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g_main_loop_quit (main_loop);
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}
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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GST_START_TEST (test_play_twice)
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{
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GstElement *bin, *src1, *src2, *adder, *sink;
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GstBus *bus;
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gboolean res;
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GstPad *srcpad;
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GstStreamConsistency *consist;
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GST_INFO ("preparing test");
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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src1 = gst_element_factory_make ("audiotestsrc", "src1");
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g_object_set (src1, "wave", 4, NULL); /* silence */
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src2 = gst_element_factory_make ("audiotestsrc", "src2");
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g_object_set (src2, "wave", 4, NULL); /* silence */
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adder = gst_element_factory_make ("adder", "adder");
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
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res = gst_element_link (src1, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (src2, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (adder, sink);
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fail_unless (res == TRUE, NULL);
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srcpad = gst_element_get_static_pad (adder, "src");
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consist = gst_consistency_checker_new (srcpad);
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gst_object_unref (srcpad);
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play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
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GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
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GST_SEEK_TYPE_SET, (GstClockTime) 0,
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GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
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play_count = 0;
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main_loop = g_main_loop_new (NULL, FALSE);
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g_signal_connect (bus, "message::segment-done",
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(GCallback) test_play_twice_message_received, bin);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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GST_INFO ("starting test");
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/* prepare playing */
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res = gst_element_set_state (bin, GST_STATE_PAUSED);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* wait for completion */
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res =
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gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
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GST_CLOCK_TIME_NONE);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
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fail_unless (res == TRUE, NULL);
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GST_INFO ("seeked");
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/* run pipeline */
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res = gst_element_set_state (bin, GST_STATE_PLAYING);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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g_main_loop_run (main_loop);
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res = gst_element_set_state (bin, GST_STATE_NULL);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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fail_unless (play_count == 2, NULL);
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/* cleanup */
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g_main_loop_unref (main_loop);
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gst_consistency_checker_free (consist);
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gst_event_ref (play_seek_event);
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gst_object_unref (G_OBJECT (bus));
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gst_object_unref (G_OBJECT (bin));
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}
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GST_END_TEST;
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GST_START_TEST (test_play_twice_then_add_and_play_again)
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{
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GstElement *bin, *src1, *src2, *src3, *adder, *sink;
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GstBus *bus;
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gboolean res;
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gint i;
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GstPad *srcpad;
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GstStreamConsistency *consist;
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GST_INFO ("preparing test");
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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src1 = gst_element_factory_make ("audiotestsrc", "src1");
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g_object_set (src1, "wave", 4, NULL); /* silence */
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src2 = gst_element_factory_make ("audiotestsrc", "src2");
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g_object_set (src2, "wave", 4, NULL); /* silence */
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adder = gst_element_factory_make ("adder", "adder");
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (bin), src1, src2, adder, sink, NULL);
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srcpad = gst_element_get_static_pad (adder, "src");
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consist = gst_consistency_checker_new (srcpad);
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gst_object_unref (srcpad);
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res = gst_element_link (src1, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (src2, adder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (adder, sink);
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fail_unless (res == TRUE, NULL);
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play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
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GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
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GST_SEEK_TYPE_SET, (GstClockTime) 0,
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GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
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main_loop = g_main_loop_new (NULL, FALSE);
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g_signal_connect (bus, "message::segment-done",
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(GCallback) test_play_twice_message_received, bin);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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/* run it twice */
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for (i = 0; i < 2; i++) {
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play_count = 0;
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GST_INFO ("starting test-loop %d", i);
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/* prepare playing */
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res = gst_element_set_state (bin, GST_STATE_PAUSED);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* wait for completion */
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res =
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gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
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GST_CLOCK_TIME_NONE);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
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fail_unless (res == TRUE, NULL);
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GST_INFO ("seeked");
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/* run pipeline */
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res = gst_element_set_state (bin, GST_STATE_PLAYING);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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g_main_loop_run (main_loop);
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res = gst_element_set_state (bin, GST_STATE_READY);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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fail_unless (play_count == 2, NULL);
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/* plug another source */
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if (i == 0) {
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src3 = gst_element_factory_make ("audiotestsrc", "src3");
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g_object_set (src3, "wave", 4, NULL); /* silence */
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gst_bin_add (GST_BIN (bin), src3);
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res = gst_element_link (src3, adder);
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fail_unless (res == TRUE, NULL);
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}
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gst_consistency_checker_reset (consist);
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}
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res = gst_element_set_state (bin, GST_STATE_NULL);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* cleanup */
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g_main_loop_unref (main_loop);
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gst_event_ref (play_seek_event);
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gst_consistency_checker_free (consist);
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gst_object_unref (G_OBJECT (bus));
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gst_object_unref (G_OBJECT (bin));
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}
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GST_END_TEST;
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static void
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test_live_seeking_eos_message_received (GstBus * bus, GstMessage * message,
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GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_EOS:
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g_main_loop_quit (main_loop);
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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/* test failing seeks on live-sources */
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GST_START_TEST (test_live_seeking)
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{
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GstElement *bin, *src1, *src2, *ac1, *ac2, *adder, *sink;
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GstBus *bus;
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gboolean res;
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GstPad *srcpad;
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gint i;
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GstStreamConsistency *consist;
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|
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 totally 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;
|
|
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");
|
|
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;
|
|
|
|
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_%u");
|
|
fail_if (pad == NULL, NULL);
|
|
|
|
srcpad = gst_element_get_static_pad (adder, "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 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)
|
|
{
|
|
GstSegment segment;
|
|
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_%u");
|
|
fail_if (sinkpad == NULL, NULL);
|
|
|
|
/* send segment to adder */
|
|
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);
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
#if G_BYTE_ORDER == G_BIG_ENDIAN
|
|
"format", G_TYPE_STRING, "S16BE",
|
|
#else
|
|
"format", G_TYPE_STRING, "S16LE",
|
|
#endif
|
|
"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);
|
|
|
|
/* 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", 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_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_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_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);
|