/* GStreamer * * Copyright (C) 2016 Igalia S.L. * @author Philippe Normand * * 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. */ #include #include #include #include static GMainLoop *main_loop; static void message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) { GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, GST_MESSAGE_SRC (message), message); switch (message->type) { case GST_MESSAGE_EOS: g_main_loop_quit (main_loop); break; case GST_MESSAGE_WARNING:{ GError *gerror; gchar *debug; gst_message_parse_warning (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); break; } case GST_MESSAGE_ERROR:{ GError *gerror; gchar *debug; gst_message_parse_error (message, &gerror, &debug); gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); g_error_free (gerror); g_free (debug); fail ("Error!"); break; } default: break; } } static void on_rtpbinreceive_pad_added (GstElement * element, GstPad * new_pad, gpointer data) { GstElement *pipeline = GST_ELEMENT (data); gchar *pad_name = gst_pad_get_name (new_pad); if (g_str_has_prefix (pad_name, "recv_rtp_src_")) { GstCaps *caps = gst_pad_get_current_caps (new_pad); GstStructure *s = gst_caps_get_structure (caps, 0); const gchar *media_type = gst_structure_get_string (s, "media"); gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type); GstElement *rtpdepayloader = gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name); GstPad *sinkpad; g_free (depayloader_name); fail_unless (rtpdepayloader != NULL, NULL); sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink"); gst_pad_link (new_pad, sinkpad); gst_object_unref (sinkpad); gst_object_unref (rtpdepayloader); gst_caps_unref (caps); } g_free (pad_name); } static guint on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data) { static gboolean create_session = FALSE; guint session_id = 0; if (create_session) { session_id = 1; } else { create_session = TRUE; /* use existing session 0, a new session will be created for the next discovered bundled SSRC */ } return session_id; } static GstCaps * on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt, gpointer user_data) { GstCaps *caps = NULL; if (pt == 96) { caps = gst_caps_from_string ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000"); } else if (pt == 100) { caps = gst_caps_from_string ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240"); } return caps; } static GstElement * create_pipeline (gboolean send) { GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder, *audio_rtppayloader, *sendrtp_udpsink, *recv_rtp_udpsrc, *send_rtcp_udpsink, *recv_rtcp_udpsrc, *sendrtcp_funnel, *sendrtp_funnel; GstElement *audio_rtpdepayloader, *audio_decoder, *audio_sink; GstElement *videosrc, *video_rtppayloader, *video_rtpdepayloader, *video_sink; gboolean res; GstPad *funnel_pad, *rtp_src_pad; GstCaps *rtpcaps; gint rtp_udp_port = 5001; gint rtcp_udp_port = 5002; pipeline = gst_pipeline_new (send ? "pipeline_send" : "pipeline_receive"); rtpbin = gst_element_factory_make ("rtpbin", send ? "rtpbin_send" : "rtpbin_receive"); g_object_set (rtpbin, "latency", 200, NULL); if (!send) { g_signal_connect (rtpbin, "on-bundled-ssrc", G_CALLBACK (on_bundled_ssrc), NULL); g_signal_connect (rtpbin, "request-pt-map", G_CALLBACK (on_request_pt_map), NULL); } g_signal_connect (rtpbin, "pad-added", G_CALLBACK (on_rtpbinreceive_pad_added), pipeline); gst_bin_add (GST_BIN (pipeline), rtpbin); if (send) { audiosrc = gst_element_factory_make ("audiotestsrc", NULL); audio_encoder = gst_element_factory_make ("alawenc", NULL); audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL); g_object_set (audio_rtppayloader, "pt", 96, NULL); g_object_set (audio_rtppayloader, "seqnum-offset", 1, NULL); videosrc = gst_element_factory_make ("videotestsrc", NULL); video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL); g_object_set (video_rtppayloader, "pt", 100, "seqnum-offset", 1, NULL); g_object_set (audiosrc, "num-buffers", 5, NULL); g_object_set (videosrc, "num-buffers", 5, NULL); /* muxed rtcp */ sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel"); send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL); g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL); g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL); g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL); g_object_set (send_rtcp_udpsink, "async", FALSE, NULL); /* outgoing bundled stream */ sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel"); sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL); g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL); g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL); gst_bin_add_many (GST_BIN (pipeline), audiosrc, audio_encoder, audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink, sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL); res = gst_element_link (audiosrc, audio_encoder); fail_unless (res == TRUE, NULL); res = gst_element_link (audio_encoder, audio_rtppayloader); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (audio_rtppayloader, "src", rtpbin, "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link (videosrc, video_rtppayloader); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (video_rtppayloader, "src", rtpbin, "send_rtp_sink_1", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (sendrtp_funnel, "src", sendrtp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u"); rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0"); res = gst_pad_link (rtp_src_pad, funnel_pad); gst_object_unref (funnel_pad); gst_object_unref (rtp_src_pad); funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u"); rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_1"); res = gst_pad_link (rtp_src_pad, funnel_pad); gst_object_unref (funnel_pad); gst_object_unref (rtp_src_pad); res = gst_element_link_pads_full (sendrtcp_funnel, "src", send_rtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u"); rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); res = gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (funnel_pad); gst_object_unref (rtp_src_pad); funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u"); rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1"); res = gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (funnel_pad); gst_object_unref (rtp_src_pad); } else { recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL); g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL); rtpcaps = gst_caps_from_string ("application/x-rtp"); g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL); gst_caps_unref (rtpcaps); recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL); g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL); audio_rtpdepayloader = gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader"); audio_decoder = gst_element_factory_make ("alawdec", NULL); audio_sink = gst_element_factory_make ("fakesink", NULL); g_object_set (audio_sink, "sync", TRUE, NULL); video_rtpdepayloader = gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader"); video_sink = gst_element_factory_make ("fakesink", NULL); g_object_set (video_sink, "sync", TRUE, NULL); gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc, audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader, video_sink, NULL); res = gst_element_link_pads_full (audio_rtpdepayloader, "src", audio_decoder, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link (audio_decoder, audio_sink); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (video_rtpdepayloader, "src", video_sink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); /* request a single receiving RTP session. */ res = gst_element_link_pads_full (recv_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (recv_rtp_udpsrc, "src", rtpbin, "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); } return pipeline; } GST_START_TEST (test_simple_rtpbin_bundle) { GstElement *send_pipeline, *recv_pipeline; GstBus *send_bus, *recv_bus; GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE; GstElement *rtpbin_receive; GObject *rtp_session; main_loop = g_main_loop_new (NULL, FALSE); send_pipeline = create_pipeline (TRUE); recv_pipeline = create_pipeline (FALSE); send_bus = gst_element_get_bus (send_pipeline); gst_bus_add_signal_watch_full (send_bus, G_PRIORITY_HIGH); g_signal_connect (send_bus, "message::error", (GCallback) message_received, send_pipeline); g_signal_connect (send_bus, "message::warning", (GCallback) message_received, send_pipeline); g_signal_connect (send_bus, "message::eos", (GCallback) message_received, send_pipeline); recv_bus = gst_element_get_bus (recv_pipeline); gst_bus_add_signal_watch_full (recv_bus, G_PRIORITY_HIGH); g_signal_connect (recv_bus, "message::error", (GCallback) message_received, recv_pipeline); g_signal_connect (recv_bus, "message::warning", (GCallback) message_received, recv_pipeline); g_signal_connect (recv_bus, "message::eos", (GCallback) message_received, recv_pipeline); state_res = gst_element_set_state (recv_pipeline, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); state_res = gst_element_set_state (send_pipeline, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); GST_INFO ("enter mainloop"); g_main_loop_run (main_loop); GST_INFO ("exit mainloop"); rtpbin_receive = gst_bin_get_by_name (GST_BIN (recv_pipeline), "rtpbin_receive"); fail_if (rtpbin_receive == NULL, NULL); /* Check that 2 RTP sessions where created while only one was explicitely requested. */ g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 0, &rtp_session); fail_if (rtp_session == NULL, NULL); g_object_unref (rtp_session); g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 1, &rtp_session); fail_if (rtp_session == NULL, NULL); g_object_unref (rtp_session); gst_object_unref (rtpbin_receive); state_res = gst_element_set_state (send_pipeline, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); state_res = gst_element_set_state (recv_pipeline, 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 (send_bus); gst_object_unref (send_bus); gst_object_unref (send_pipeline); gst_bus_remove_signal_watch (recv_bus); gst_object_unref (recv_bus); gst_object_unref (recv_pipeline); } GST_END_TEST; static Suite * rtpbundle_suite (void) { Suite *s = suite_create ("rtpbundle"); TCase *tc_chain = tcase_create ("general"); tcase_set_timeout (tc_chain, 10000); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_simple_rtpbin_bundle); return s; } GST_CHECK_MAIN (rtpbundle);