| /* GStreamer |
| * |
| * unit test for audiomixer |
| * |
| * Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org> |
| * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include <config.h> |
| #endif |
| |
| #ifdef HAVE_VALGRIND |
| # include <valgrind/valgrind.h> |
| #endif |
| |
| #include <unistd.h> |
| |
| #include <gst/check/gstcheck.h> |
| #include <gst/check/gstconsistencychecker.h> |
| #include <gst/audio/audio.h> |
| #include <gst/base/gstbasesrc.h> |
| #include <gst/controller/gstdirectcontrolbinding.h> |
| #include <gst/controller/gstinterpolationcontrolsource.h> |
| |
| static GMainLoop *main_loop; |
| |
| /* 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; |
| |
| main_loop = g_main_loop_new (NULL, FALSE); |
| g_signal_connect (bus, "message::segment-done", |
| (GCallback) test_event_message_received, bin); |
| g_signal_connect (bus, "message::error", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); |
| |
| GST_INFO ("starting test"); |
| |
| /* prepare playing */ |
| 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 */ |
| g_main_loop_unref (main_loop); |
| 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; |
| |
| main_loop = g_main_loop_new (NULL, FALSE); |
| g_signal_connect (bus, "message::segment-done", |
| (GCallback) test_play_twice_message_received, bin); |
| g_signal_connect (bus, "message::error", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); |
| |
| GST_INFO ("starting test"); |
| |
| /* prepare playing */ |
| 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 */ |
| g_main_loop_unref (main_loop); |
| 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); |
| |
| main_loop = g_main_loop_new (NULL, FALSE); |
| g_signal_connect (bus, "message::segment-done", |
| (GCallback) test_play_twice_message_received, bin); |
| g_signal_connect (bus, "message::error", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); |
| g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); |
| |
| /* run it twice */ |
| for (i = 0; i < 2; i++) { |
| play_count = 0; |
| |
| GST_INFO ("starting test-loop %d", i); |
| |
| /* prepare playing */ |
| 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 */ |
| g_main_loop_unref (main_loop); |
| 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"); |
| 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); |
| |
| 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); |
| |
| 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 (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 (main_loop) |
| g_main_loop_unref (main_loop); |
| if (play_seek_event) |
| gst_event_unref (play_seek_event); |
| 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); |
| |
| 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 */ |
| 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 */ |
| g_main_loop_unref (main_loop); |
| 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); |
| |
| 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 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 */ |
| g_main_loop_unref (main_loop); |
| 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 DURATION is %" GST_TIME_FORMAT, |
| gst_buffer_get_size (buffer), 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"); |
| 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_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); |
| 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 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); |
| 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_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); |
| |
| 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); |
| |
| main_loop = g_main_loop_new (NULL, FALSE); |
| 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 */ |
| g_main_loop_unref (main_loop); |
| 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"); |
| |
| main_loop = g_main_loop_new (NULL, FALSE); |
| |
| /* 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); |
| g_main_loop_unref (main_loop); |
| } |
| |
| 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); |
| |
| /* 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); |