/* GStreamer * * unit test for audiomixer * * Copyright (C) 2005 Thomas Vander Stichele * Copyright (C) 2013 Sebastian Dröge * * 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 #endif #ifdef HAVE_VALGRIND # include #endif #include #include #include #include #include #include #include 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; } /* make sure downstream gets a CAPS event before buffers are sent */ GST_START_TEST (test_caps) { GstElement *pipeline, *src, *audiomixer, *sink; GstStateChangeReturn state_res; GstCaps *caps; GstPad *pad; /* build pipeline */ pipeline = gst_pipeline_new ("pipeline"); src = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (pipeline), src, audiomixer, sink, NULL); fail_unless (gst_element_link_many (src, audiomixer, sink, NULL)); /* prepare playing */ state_res = gst_element_set_state (pipeline, GST_STATE_PAUSED); fail_unless_equals_int (state_res, GST_STATE_CHANGE_ASYNC); /* wait for preroll */ state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE); fail_unless_equals_int (state_res, GST_STATE_CHANGE_SUCCESS); /* check caps on fakesink */ pad = gst_element_get_static_pad (sink, "sink"); caps = gst_pad_get_current_caps (pad); fail_unless (caps != NULL); gst_caps_unref (caps); gst_object_unref (pad); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } GST_END_TEST; /* check that caps set on the property are honoured */ GST_START_TEST (test_filter_caps) { GstElement *pipeline, *src, *audiomixer, *sink; GstStateChangeReturn state_res; GstCaps *filter_caps, *caps; GstPad *pad; filter_caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, GST_AUDIO_NE (F32), "layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, "channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL); /* build pipeline */ pipeline = gst_pipeline_new ("pipeline"); src = gst_element_factory_make ("audiotestsrc", NULL); g_object_set (src, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", NULL); g_object_set (audiomixer, "caps", filter_caps, NULL); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (pipeline), src, audiomixer, sink, NULL); fail_unless (gst_element_link_many (src, audiomixer, sink, NULL)); /* prepare playing */ state_res = gst_element_set_state (pipeline, GST_STATE_PAUSED); fail_unless_equals_int (state_res, GST_STATE_CHANGE_ASYNC); /* wait for preroll */ state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE); fail_unless_equals_int (state_res, GST_STATE_CHANGE_SUCCESS); /* check caps on fakesink */ pad = gst_element_get_static_pad (sink, "sink"); caps = gst_pad_get_current_caps (pad); fail_unless (caps != NULL); GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps); fail_unless (gst_caps_is_equal_fixed (caps, filter_caps)); gst_caps_unref (caps); gst_object_unref (pad); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); gst_caps_unref (filter_caps); } GST_END_TEST; static gboolean set_playing (GstElement * element) { GstStateChangeReturn state_res; state_res = gst_element_set_state (element, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); return FALSE; } static void message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) { GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); switch (message->type) { case GST_MESSAGE_EOS: g_main_loop_quit (main_loop); break; case GST_MESSAGE_WARNING:{ GError *gerror; gchar *debug; gst_message_parse_warning (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); break; } case GST_MESSAGE_ERROR:{ GError *gerror; gchar *debug; gst_message_parse_error (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); g_main_loop_quit (main_loop); break; } default: break; } } static GstFormat format = GST_FORMAT_UNDEFINED; static gint64 position = -1; static void test_event_message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) { GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); switch (message->type) { case GST_MESSAGE_SEGMENT_DONE: gst_message_parse_segment_done (message, &format, &position); GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position); g_main_loop_quit (main_loop); break; default: g_assert_not_reached (); break; } } GST_START_TEST (test_event) { GstElement *bin, *src1, *src2, *audiomixer, *sink; GstBus *bus; GstEvent *seek_event; GstStateChangeReturn state_res; gboolean res; GstPad *srcpad, *sinkpad; GstStreamConsistency *chk_1, *chk_2, *chk_3; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL); res = gst_element_link (src1, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (src2, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); srcpad = gst_element_get_static_pad (audiomixer, "src"); chk_3 = gst_consistency_checker_new (srcpad); gst_object_unref (srcpad); /* create consistency checkers for the pads */ srcpad = gst_element_get_static_pad (src1, "src"); chk_1 = gst_consistency_checker_new (srcpad); sinkpad = gst_pad_get_peer (srcpad); gst_consistency_checker_add_pad (chk_3, sinkpad); gst_object_unref (sinkpad); gst_object_unref (srcpad); srcpad = gst_element_get_static_pad (src2, "src"); chk_2 = gst_consistency_checker_new (srcpad); sinkpad = gst_pad_get_peer (srcpad); gst_consistency_checker_add_pad (chk_3, sinkpad); gst_object_unref (sinkpad); gst_object_unref (srcpad); seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); format = GST_FORMAT_UNDEFINED; position = -1; g_signal_connect (bus, "message::segment-done", (GCallback) test_event_message_received, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); GST_INFO ("starting test"); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (bin, seek_event); fail_unless (res == TRUE, NULL); /* run pipeline */ g_idle_add ((GSourceFunc) set_playing, bin); GST_INFO ("running main loop"); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); ck_assert_int_eq (position, 2 * GST_SECOND); /* cleanup */ gst_consistency_checker_free (chk_1); gst_consistency_checker_free (chk_2); gst_consistency_checker_free (chk_3); gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (bin); } GST_END_TEST; static guint play_count = 0; static GstEvent *play_seek_event = NULL; static void test_play_twice_message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) { gboolean res; GstStateChangeReturn state_res; GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); switch (message->type) { case GST_MESSAGE_SEGMENT_DONE: play_count++; if (play_count == 1) { state_res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_READY); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* prepare playing again */ state_res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (GST_ELEMENT (bin), gst_event_ref (play_seek_event)); fail_unless (res == TRUE, NULL); state_res = gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); } else { g_main_loop_quit (main_loop); } break; default: g_assert_not_reached (); break; } } GST_START_TEST (test_play_twice) { GstElement *bin, *src1, *src2, *audiomixer, *sink; GstBus *bus; gboolean res; GstStateChangeReturn state_res; GstPad *srcpad; GstStreamConsistency *consist; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL); res = gst_element_link (src1, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (src2, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); srcpad = gst_element_get_static_pad (audiomixer, "src"); consist = gst_consistency_checker_new (srcpad); gst_object_unref (srcpad); play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); play_count = 0; g_signal_connect (bus, "message::segment-done", (GCallback) test_play_twice_message_received, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); GST_INFO ("starting test"); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); fail_unless (res == TRUE, NULL); GST_INFO ("seeked"); /* run pipeline */ g_idle_add ((GSourceFunc) set_playing, bin); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); ck_assert_int_eq (play_count, 2); /* cleanup */ 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, *src1, *src2, *src3, *audiomixer, *sink; GstBus *bus; gboolean res; GstStateChangeReturn state_res; gint i; GstPad *srcpad; GstStreamConsistency *consist; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL); srcpad = gst_element_get_static_pad (audiomixer, "src"); consist = gst_consistency_checker_new (srcpad); gst_object_unref (srcpad); res = gst_element_link (src1, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (src2, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); g_signal_connect (bus, "message::segment-done", (GCallback) test_play_twice_message_received, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); /* run it twice */ for (i = 0; i < 2; i++) { play_count = 0; GST_INFO ("starting test-loop %d", i); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); fail_unless (res == TRUE, NULL); GST_INFO ("seeked"); /* run pipeline */ g_idle_add ((GSourceFunc) set_playing, bin); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_READY); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); ck_assert_int_eq (play_count, 2); /* plug another source */ if (i == 0) { src3 = gst_element_factory_make ("audiotestsrc", "src3"); g_object_set (src3, "wave", 4, NULL); /* silence */ gst_bin_add (GST_BIN (bin), src3); res = gst_element_link (src3, audiomixer); fail_unless (res == TRUE, NULL); } gst_consistency_checker_reset (consist); } state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* cleanup */ 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 void test_live_seeking_eos_message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) { GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); switch (message->type) { case GST_MESSAGE_EOS: g_main_loop_quit (main_loop); break; default: g_assert_not_reached (); break; } } static GstElement * test_live_seeking_try_audiosrc (const gchar * factory_name) { GstElement *src; GstStateChangeReturn state_res; if (!(src = gst_element_factory_make (factory_name, NULL))) { GST_INFO ("can't make '%s', skipping", factory_name); return NULL; } /* Test that the audio source can get to ready, else skip */ state_res = gst_element_set_state (src, GST_STATE_READY); gst_element_set_state (src, GST_STATE_NULL); if (state_res == GST_STATE_CHANGE_FAILURE) { GST_INFO_OBJECT (src, "can't go to ready, skipping"); gst_object_unref (src); return NULL; } return src; } /* test failing seeks on live-sources */ GST_START_TEST (test_live_seeking) { GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink; GstCaps *caps; GstBus *bus; gboolean res; GstPad *srcpad; GstPad *sinkpad; gint i; GstStateChangeReturn state_res; GstStreamConsistency *consist; /* don't use autoaudiosrc, as then we can't set anything here */ const gchar *audio_src_factories[] = { "alsasrc", "pulseaudiosrc" }; GST_INFO ("preparing test"); play_seek_event = NULL; /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) { src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]); } if (!src1) { /* normal audiosources behave differently than audiotestsrc */ src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, "is-live", TRUE, NULL); /* silence */ } else { /* live sources ignore seeks, force eos after 2 sec (4 buffers half second * each) */ g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL); } audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); cf = gst_element_factory_make ("capsfilter", "capsfilter"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL); res = gst_element_link (src1, cf); fail_unless (res == TRUE, NULL); res = gst_element_link (cf, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); gst_element_set_state (bin, GST_STATE_PLAYING); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); sinkpad = gst_element_get_static_pad (sink, "sink"); fail_unless (sinkpad != NULL); caps = gst_pad_get_current_caps (sinkpad); fail_unless (caps != NULL); gst_object_unref (sinkpad); gst_element_set_state (bin, GST_STATE_NULL); g_object_set (cf, "caps", caps, NULL); src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ gst_bin_add (GST_BIN (bin), src2); res = gst_element_link_filtered (src2, audiomixer, caps); fail_unless (res == TRUE, NULL); gst_caps_unref (caps); play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) test_live_seeking_eos_message_received, bin); srcpad = gst_element_get_static_pad (audiomixer, "src"); consist = gst_consistency_checker_new (srcpad); gst_object_unref (srcpad); GST_INFO ("starting test"); /* run it twice */ for (i = 0; i < 2; i++) { GST_INFO ("starting test-loop %d", i); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); fail_unless (res == TRUE, NULL); GST_INFO ("seeked"); /* run pipeline */ g_idle_add ((GSourceFunc) set_playing, bin); GST_INFO ("playing"); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); gst_consistency_checker_reset (consist); } /* cleanup */ GST_INFO ("cleaning up"); gst_consistency_checker_free (consist); if (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, NULL); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); /* one buffer less, we connect with 1 buffer of delay */ g_object_set (src2, "num-buffers", 3, NULL); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL); res = gst_element_link (src1, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); srcpad = gst_element_get_static_pad (audiomixer, "src"); gst_object_unref (srcpad); g_signal_connect (bus, "message::segment-done", (GCallback) message_received, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); GST_INFO ("starting test"); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* add other element */ gst_bin_add_many (GST_BIN (bin), src2, NULL); /* now link the second element */ res = gst_element_link (src2, audiomixer); fail_unless (res == TRUE, NULL); /* set to PAUSED as well */ state_res = gst_element_set_state (src2, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* now play all */ g_idle_add ((GSourceFunc) set_playing, bin); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* cleanup */ gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (bin); } GST_END_TEST; /* check if removing pads work as expected */ GST_START_TEST (test_remove_pad) { GstElement *bin, *src, *audiomixer, *sink; GstBus *bus; GstPad *pad, *srcpad; gboolean res; GstStateChangeReturn state_res; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); src = gst_element_factory_make ("audiotestsrc", "src"); g_object_set (src, "num-buffers", 4, NULL); g_object_set (src, "wave", 4, NULL); audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL); res = gst_element_link (src, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); /* create an unconnected sinkpad in audiomixer */ pad = gst_element_get_request_pad (audiomixer, "sink_%u"); fail_if (pad == NULL, NULL); srcpad = gst_element_get_static_pad (audiomixer, "src"); gst_object_unref (srcpad); g_signal_connect (bus, "message::segment-done", (GCallback) message_received, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); GST_INFO ("starting test"); /* prepare playing, this will not preroll as audiomixer is waiting * on the unconnected sinkpad. */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion for one second, will return ASYNC */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND); ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC); /* get rid of the pad now, audiomixer should stop waiting on it and * continue the preroll */ gst_element_release_request_pad (audiomixer, pad); gst_object_unref (pad); /* wait for completion, should work now */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* now play all */ g_idle_add ((GSourceFunc) set_playing, bin); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* cleanup */ 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 = gst_buffer_new_and_alloc (44100); GST_BUFFER_TIMESTAMP (buffer) = 0; GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND; GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); ret = gst_pad_chain (sinkpad, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); /* The aggregation is done in a dedicated thread, so we can't * know when it is actually going to happen, so we use a DRAIN query * to wait for it to complete. */ gst_pad_query (sinkpad, drain); fail_unless (handoff_buffer == NULL); /* should be partially clipped */ buffer = gst_buffer_new_and_alloc (44100); GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND; GST_BUFFER_FLAG_SET (buffer, 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 = gst_buffer_new_and_alloc (44100); GST_BUFFER_TIMESTAMP (buffer) = 1150 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND; GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); ret = gst_pad_chain (sinkpad, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_query (sinkpad, drain); fail_unless (handoff_buffer != NULL); 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 = gst_buffer_new_and_alloc (44100); GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND; GST_BUFFER_FLAG_SET (buffer, 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 */ state_res = gst_element_set_state (bin, GST_STATE_PLAYING); fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration); fail_unless (res, NULL); ck_assert_int_eq (duration, 3000); gst_element_set_state (bin, GST_STATE_NULL); gst_object_unref (bin); } GST_END_TEST; GST_START_TEST (test_duration_unknown_overrides) { GstElement *bin, *src[3], *audiomixer, *sink; GstStateChangeReturn state_res; GstFormat format = GST_FORMAT_TIME; gboolean res; gint64 duration; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); /* 3 sources, an audiomixer and a fakesink */ src[0] = gst_element_factory_make ("audiotestsrc", NULL); src[1] = gst_element_factory_make ("audiotestsrc", NULL); src[2] = gst_element_factory_make ("audiotestsrc", NULL); audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink, NULL); gst_element_link (src[0], audiomixer); gst_element_link (src[1], audiomixer); gst_element_link (src[2], audiomixer); gst_element_link (audiomixer, sink); /* irks, duration is reset on basesrc */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); /* set durations on src */ GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE; GST_BASE_SRC (src[1])->segment.duration = 3000; GST_BASE_SRC (src[2])->segment.duration = 2000; /* set to playing */ state_res = gst_element_set_state (bin, GST_STATE_PLAYING); fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); /* wait for completion */ state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_CLOCK_TIME_NONE); fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration); fail_unless (res, NULL); ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE); gst_element_set_state (bin, GST_STATE_NULL); gst_object_unref (bin); } GST_END_TEST; static gboolean looped = FALSE; static void loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin) { GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); if (looped) { g_main_loop_quit (main_loop); } else { GstEvent *seek_event; gboolean res; seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_SEGMENT, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND); res = gst_element_send_event (bin, seek_event); fail_unless (res == TRUE, NULL); looped = TRUE; } } GST_START_TEST (test_loop) { GstElement *bin, *src1, *src2, *audiomixer, *sink; GstBus *bus; GstEvent *seek_event; GstStateChangeReturn state_res; gboolean res; GST_INFO ("preparing test"); /* build pipeline */ bin = gst_pipeline_new ("pipeline"); bus = gst_element_get_bus (bin); gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL); res = gst_element_link (src1, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (src2, audiomixer); fail_unless (res == TRUE, NULL); res = gst_element_link (audiomixer, sink); fail_unless (res == TRUE, NULL); seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, (GstClockTime) 0, GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND); g_signal_connect (bus, "message::segment-done", (GCallback) loop_segment_done, bin); g_signal_connect (bus, "message::error", (GCallback) message_received, bin); g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); GST_INFO ("starting test"); /* prepare playing */ state_res = gst_element_set_state (bin, GST_STATE_PAUSED); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); /* wait for completion */ state_res = gst_element_get_state (bin, NULL, NULL, GST_CLOCK_TIME_NONE); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); res = gst_element_send_event (bin, seek_event); fail_unless (res == TRUE, NULL); /* run pipeline */ g_idle_add ((GSourceFunc) set_playing, bin); GST_INFO ("running main loop"); g_main_loop_run (main_loop); state_res = gst_element_set_state (bin, GST_STATE_NULL); /* cleanup */ gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (bin); } GST_END_TEST; GST_START_TEST (test_flush_start_flush_stop) { GstPadTemplate *sink_template; GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src; GstElement *pipeline, *src1, *src2, *audiomixer, *sink; GST_INFO ("preparing test"); /* build pipeline */ pipeline = gst_pipeline_new ("pipeline"); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "wave", 4, NULL); /* silence */ src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "wave", 4, NULL); /* silence */ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL); sink_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer), "sink_%u"); fail_unless (GST_IS_PAD_TEMPLATE (sink_template)); sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL); srcpad1 = gst_element_get_static_pad (src1, "src"); gst_pad_link (srcpad1, sinkpad1); sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL); tmppad = gst_element_get_static_pad (src2, "src"); gst_pad_link (tmppad, sinkpad2); gst_object_unref (tmppad); gst_element_link (audiomixer, sink); gst_element_set_state (pipeline, GST_STATE_PLAYING); fail_unless (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) == GST_STATE_CHANGE_SUCCESS); audiomixer_src = gst_element_get_static_pad (audiomixer, "src"); fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); gst_pad_send_event (sinkpad1, gst_event_new_flush_start ()); fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); fail_unless (GST_PAD_IS_FLUSHING (sinkpad1)); /* Hold the streamlock to make sure the flush stop is not between the attempted push of a segment event and of the following buffer. */ GST_PAD_STREAM_LOCK (srcpad1); gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE)); GST_PAD_STREAM_UNLOCK (srcpad1); fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); fail_if (GST_PAD_IS_FLUSHING (sinkpad1)); gst_object_unref (audiomixer_src); gst_element_release_request_pad (audiomixer, sinkpad1); gst_object_unref (sinkpad1); gst_element_release_request_pad (audiomixer, sinkpad2); gst_object_unref (sinkpad2); gst_object_unref (srcpad1); /* cleanup */ gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } GST_END_TEST; static void handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad, gpointer user_data) { GList **received_buffers = user_data; GST_DEBUG ("got buffer %p", buffer); *received_buffers = g_list_append (*received_buffers, gst_buffer_ref (buffer)); } typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2); typedef void (*CheckBuffersFunction) (GList * buffers); static void run_sync_test (SendBuffersFunction send_buffers, CheckBuffersFunction check_buffers) { GstSegment segment; GstElement *bin, *audiomixer, *queue1, *queue2, *sink; GstBus *bus; GstPad *sinkpad1, *sinkpad2; GstPad *queue1_sinkpad, *queue2_sinkpad; GstPad *pad; gboolean res; GstStateChangeReturn state_res; GstEvent *event; GstCaps *caps; GList *received_buffers = NULL; GST_INFO ("preparing test"); /* 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; GstMapInfo map; GstFlowReturn ret; buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad1, gst_event_new_eos ()); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad2, gst_event_new_eos ()); } static void check_buffers_sync (GList * received_buffers) { GstBuffer *buffer; GList *l; gint i; GstMapInfo map; /* Should have 8 * 0.5s buffers */ fail_unless_equals_int (g_list_length (received_buffers), 8); for (i = 0, l = received_buffers; l; l = l->next, i++) { buffer = l->data; gst_buffer_map (buffer, &map, GST_MAP_READ); if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { fail_unless (map.data[0] == 0); fail_unless (map.data[map.size - 1] == 0); } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { fail_unless (map.data[0] == 0); fail_unless (map.data[map.size - 1] == 0); } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[map.size - 1] == 3); } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[map.size - 1] == 3); } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { fail_unless (map.data[0] == 2); fail_unless (map.data[map.size - 1] == 2); } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { fail_unless (map.data[0] == 2); fail_unless (map.data[map.size - 1] == 2); } else { g_assert_not_reached (); } gst_buffer_unmap (buffer, &map); } } GST_START_TEST (test_sync) { run_sync_test (send_buffers_sync, check_buffers_sync); } GST_END_TEST; static void send_buffers_sync_discont (GstPad * pad1, GstPad * pad2) { GstBuffer *buffer; GstMapInfo map; GstFlowReturn ret; buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad1, gst_event_new_eos ()); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad2, gst_event_new_eos ()); } static void check_buffers_sync_discont (GList * received_buffers) { GstBuffer *buffer; GList *l; gint i; GstMapInfo map; /* Should have 8 * 0.5s buffers */ fail_unless_equals_int (g_list_length (received_buffers), 8); for (i = 0, l = received_buffers; l; l = l->next, i++) { buffer = l->data; gst_buffer_map (buffer, &map, GST_MAP_READ); if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { fail_unless (map.data[0] == 0); fail_unless (map.data[map.size - 1] == 0); } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { fail_unless (map.data[0] == 0); fail_unless (map.data[map.size - 1] == 0); } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { fail_unless (map.data[0] == 2); fail_unless (map.data[map.size - 1] == 2); } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { fail_unless (map.data[0] == 2); fail_unless (map.data[map.size - 1] == 2); } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[map.size - 1] == 3); } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[map.size - 1] == 3); } else { g_assert_not_reached (); } gst_buffer_unmap (buffer, &map); } } GST_START_TEST (test_sync_discont) { run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont); } GST_END_TEST; static void send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2) { GstBuffer *buffer; GstMapInfo map; GstFlowReturn ret; buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 750 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 1, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 1750 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad1, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad1, gst_event_new_eos ()); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 1750 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); buffer = gst_buffer_new_and_alloc (2000); gst_buffer_map (buffer, &map, GST_MAP_WRITE); memset (map.data, 2, map.size); gst_buffer_unmap (buffer, &map); GST_BUFFER_TIMESTAMP (buffer) = 2750 * GST_MSECOND; GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND; GST_DEBUG ("pushing buffer %p", buffer); ret = gst_pad_chain (pad2, buffer); ck_assert_int_eq (ret, GST_FLOW_OK); gst_pad_send_event (pad2, gst_event_new_eos ()); } static void check_buffers_sync_unaligned (GList * received_buffers) { GstBuffer *buffer; GList *l; gint i; GstMapInfo map; /* Should have 8 * 0.5s buffers */ fail_unless_equals_int (g_list_length (received_buffers), 8); for (i = 0, l = received_buffers; l; l = l->next, i++) { buffer = l->data; gst_buffer_map (buffer, &map, GST_MAP_READ); if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { fail_unless (map.data[0] == 0); fail_unless (map.data[map.size - 1] == 0); } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { fail_unless (map.data[0] == 0); fail_unless (map.data[499] == 0); fail_unless (map.data[500] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[map.size - 1] == 1); } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { fail_unless (map.data[0] == 1); fail_unless (map.data[499] == 1); fail_unless (map.data[500] == 3); fail_unless (map.data[map.size - 1] == 3); } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[499] == 3); fail_unless (map.data[500] == 3); fail_unless (map.data[map.size - 1] == 3); } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { fail_unless (map.data[0] == 3); fail_unless (map.data[499] == 3); fail_unless (map.data[500] == 2); fail_unless (map.data[map.size - 1] == 2); } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { fail_unless (map.data[0] == 2); fail_unless (map.data[499] == 2); fail_unless (map.data[500] == 2); fail_unless (map.data[map.size - 1] == 2); } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { fail_unless (map.size == 500); fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND); fail_unless (map.data[0] == 2); fail_unless (map.data[499] == 2); } else { g_assert_not_reached (); } gst_buffer_unmap (buffer, &map); } } GST_START_TEST (test_sync_unaligned) { run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned); } GST_END_TEST; GST_START_TEST (test_segment_base_handling) { GstElement *pipeline, *sink, *mix, *src1, *src2; GstPad *srcpad, *sinkpad; GstClockTime end_time; GstSample *last_sample = NULL; GstSample *sample; GstBuffer *buf; GstCaps *caps; caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL); pipeline = gst_pipeline_new ("pipeline"); mix = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("appsink", "sink"); g_object_set (sink, "caps", caps, "sync", FALSE, NULL); gst_caps_unref (caps); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL); src2 = gst_element_factory_make ("audiotestsrc", "src2"); g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL); gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL); fail_unless (gst_element_link (mix, sink)); srcpad = gst_element_get_static_pad (src1, "src"); sinkpad = gst_element_get_request_pad (mix, "sink_1"); fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); gst_object_unref (sinkpad); gst_object_unref (srcpad); srcpad = gst_element_get_static_pad (src2, "src"); sinkpad = gst_element_get_request_pad (mix, "sink_2"); fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); gst_pad_set_offset (sinkpad, 5 * GST_SECOND); gst_object_unref (sinkpad); gst_object_unref (srcpad); gst_element_set_state (pipeline, GST_STATE_PLAYING); do { g_signal_emit_by_name (sink, "pull-sample", &sample); if (sample == NULL) break; if (last_sample) gst_sample_unref (last_sample); last_sample = sample; } while (TRUE); buf = gst_sample_get_buffer (last_sample); end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); fail_unless_equals_int64 (end_time, 10 * GST_SECOND); gst_sample_unref (last_sample); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } GST_END_TEST; static void set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value, GstClockTime end, gdouble end_value) { GstControlSource *cs; GstTimedValueControlSource *tvcs; cs = gst_interpolation_control_source_new (); fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad), gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad), "volume", cs))); /* set volume interpolation mode */ g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL); tvcs = (GstTimedValueControlSource *) cs; fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value)); fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value)); gst_object_unref (cs); } GST_START_TEST (test_sinkpad_property_controller) { GstBus *bus; GstMessage *msg; GstElement *pipeline, *sink, *mix, *src1; GstPad *srcpad, *sinkpad; GError *error = NULL; gchar *debug; pipeline = gst_pipeline_new ("pipeline"); mix = gst_element_factory_make ("audiomixer", "audiomixer"); sink = gst_element_factory_make ("fakesink", "sink"); src1 = gst_element_factory_make ("audiotestsrc", "src1"); g_object_set (src1, "num-buffers", 100, NULL); gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL); fail_unless (gst_element_link (mix, sink)); srcpad = gst_element_get_static_pad (src1, "src"); sinkpad = gst_element_get_request_pad (mix, "sink_0"); fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0); gst_object_unref (sinkpad); gst_object_unref (srcpad); gst_element_set_state (pipeline, GST_STATE_PLAYING); bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_EOS | GST_MESSAGE_ERROR); switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_ERROR: gst_message_parse_error (msg, &error, &debug); g_printerr ("ERROR from element %s: %s\n", GST_OBJECT_NAME (msg->src), error->message); g_printerr ("Debug info: %s\n", debug); g_error_free (error); g_free (debug); break; case GST_MESSAGE_EOS: break; default: g_assert_not_reached (); } gst_message_unref (msg); g_object_unref (bus); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } GST_END_TEST; static Suite * audiomixer_suite (void) { Suite *s = suite_create ("audiomixer"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_caps); tcase_add_test (tc_chain, test_filter_caps); tcase_add_test (tc_chain, test_event); tcase_add_test (tc_chain, test_play_twice); tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again); tcase_add_test (tc_chain, test_live_seeking); tcase_add_test (tc_chain, test_add_pad); tcase_add_test (tc_chain, test_remove_pad); tcase_add_test (tc_chain, test_clip); tcase_add_test (tc_chain, test_duration_is_max); tcase_add_test (tc_chain, test_duration_unknown_overrides); tcase_add_test (tc_chain, test_loop); tcase_add_test (tc_chain, test_flush_start_flush_stop); tcase_add_test (tc_chain, test_sync); tcase_add_test (tc_chain, test_sync_discont); tcase_add_test (tc_chain, test_sync_unaligned); tcase_add_test (tc_chain, test_segment_base_handling); tcase_add_test (tc_chain, test_sinkpad_property_controller); tcase_add_checked_fixture (tc_chain, test_setup, test_teardown); /* 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);