/* GStreamer unit tests for flvmux * * Copyright (C) 2009 Tim-Philipp Müller * * 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 #endif #include #include static GstBusSyncReply error_cb (GstBus * bus, GstMessage * msg, gpointer user_data) { if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) { GError *err = NULL; gchar *dbg = NULL; gst_message_parse_error (msg, &err, &dbg); g_error ("ERROR: %s\n%s\n", err->message, dbg); } return GST_BUS_PASS; } static void handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad, gint * p_counter) { *p_counter += 1; GST_LOG ("counter = %d", *p_counter); } static void mux_pcm_audio (guint num_buffers, guint repeat) { GstElement *src, *sink, *flvmux, *conv, *pipeline; GstPad *sinkpad, *srcpad; gint counter; GST_LOG ("num_buffers = %u", num_buffers); pipeline = gst_pipeline_new ("pipeline"); fail_unless (pipeline != NULL, "Failed to create pipeline!"); /* kids, don't use a sync handler for this at home, really; we do because * we just want to abort and nothing else */ gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL); src = gst_element_factory_make ("audiotestsrc", "audiotestsrc"); fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!"); g_object_set (src, "num-buffers", num_buffers, NULL); conv = gst_element_factory_make ("audioconvert", "audioconvert"); fail_unless (conv != NULL, "Failed to create 'audioconvert' element!"); flvmux = gst_element_factory_make ("flvmux", "flvmux"); fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!"); sink = gst_element_factory_make ("fakesink", "fakesink"); fail_unless (sink != NULL, "Failed to create 'fakesink' element!"); g_object_set (sink, "signal-handoffs", TRUE, NULL); g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter); gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL); fail_unless (gst_element_link (src, conv)); fail_unless (gst_element_link (flvmux, sink)); /* now link the elements */ sinkpad = gst_element_get_request_pad (flvmux, "audio"); fail_unless (sinkpad != NULL, "Could not get audio request pad"); srcpad = gst_element_get_static_pad (conv, "src"); fail_unless (srcpad != NULL, "Could not get audioconvert's source pad"); fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); do { GstStateChangeReturn state_ret; GstMessage *msg; GST_LOG ("repeat=%d", repeat); counter = 0; state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED); fail_unless (state_ret != GST_STATE_CHANGE_FAILURE); if (state_ret == GST_STATE_CHANGE_ASYNC) { GST_LOG ("waiting for pipeline to reach PAUSED state"); state_ret = gst_element_get_state (pipeline, NULL, NULL, -1); fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS); } GST_LOG ("PAUSED, let's do the rest of it"); state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING); fail_unless (state_ret != GST_STATE_CHANGE_FAILURE); msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); fail_unless (msg != NULL, "Expected EOS message on bus!"); GST_LOG ("EOS"); gst_message_unref (msg); /* should have some output */ fail_unless (counter > 2); fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL), GST_STATE_CHANGE_SUCCESS); /* repeat = test re-usability */ --repeat; } while (repeat > 0); gst_object_unref (pipeline); } GST_START_TEST (test_index_writing) { /* note: there's a magic 128 value in flvmux when doing index writing */ if ((__i__ % 33) == 1) mux_pcm_audio (__i__, 2); } GST_END_TEST; static Suite * flvmux_suite (void) { Suite *s = suite_create ("flvmux"); TCase *tc_chain = tcase_create ("general"); gint loop = 499; suite_add_tcase (s, tc_chain); #ifdef HAVE_VALGRIND if (RUNNING_ON_VALGRIND) { loop = 140; } #endif tcase_add_loop_test (tc_chain, test_index_writing, 1, loop); return s; } GST_CHECK_MAIN (flvmux)