/* GStreamer * * Copyright (C) 2013 Collabora Ltd. * @author Julien Isorce * * 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 GstPad *srcpad; static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtcp") ); static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtcp") ); 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 GstBuffer * create_rtcp_app (guint32 ssrc) { GInetAddress *inet_addr_0 = g_inet_address_new_from_string ("192.168.1.1"); guint16 port = 5678; GSocketAddress *socket_addr_0 = g_inet_socket_address_new (inet_addr_0, port); GstBuffer *old_rtcp_buffer = gst_rtcp_buffer_new (1400); GstBuffer *rtcp_buffer = NULL; GstRTCPPacket *rtcp_packet = NULL; GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT; /* make sure to have a writable buffer before * changing anything */ rtcp_buffer = gst_buffer_make_writable (old_rtcp_buffer); if (old_rtcp_buffer != rtcp_buffer) { gst_buffer_unref (old_rtcp_buffer); } gst_buffer_add_net_address_meta (rtcp_buffer, socket_addr_0); /* need to begin with rr */ gst_rtcp_buffer_map (rtcp_buffer, GST_MAP_READWRITE, &rtcp); rtcp_packet = g_slice_new0 (GstRTCPPacket); gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_RR, rtcp_packet); gst_rtcp_packet_rr_set_ssrc (rtcp_packet, ssrc); /* useful to make the rtcp buffer valid */ rtcp_packet = g_slice_new0 (GstRTCPPacket); gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_APP, rtcp_packet); gst_rtcp_buffer_unmap (&rtcp); return rtcp_buffer; } static guint nb_ssrc_changes; static guint ssrc_prev; static GstPadProbeReturn rtpsession_sinkpad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstPadProbeReturn ret = GST_PAD_PROBE_OK; if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) { GstBuffer *buffer = GST_BUFFER (info->data); GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; GstBuffer *rtcp_buffer = 0; guint ssrc = 0; /* retrieve current ssrc */ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp); ssrc = gst_rtp_buffer_get_ssrc (&rtp); gst_rtp_buffer_unmap (&rtp); /* if not first buffer, check that our ssrc has changed */ if (ssrc_prev != -1 && ssrc != ssrc_prev) ++nb_ssrc_changes; /* update prev ssrc */ ssrc_prev = ssrc; /* feint a collision on recv_rtcp_sink pad of gstrtpsession * (note that after being marked as collied the rtpsession ignores * all non bye packets) */ rtcp_buffer = create_rtcp_app (ssrc); /* push collied packet on recv_rtcp_sink */ gst_pad_push (srcpad, rtcp_buffer); } return ret; } static GstFlowReturn fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer) { return GST_FLOW_OK; } /* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ * rtpsession ! fakesink * It manually pushs buffer into rtpsession with same ssrc but different * ip so that collision can be detected * The test checks that the payloader change their ssrc */ GST_START_TEST (test_master_ssrc_collision) { GstElement *bin, *src, *encoder, *rtppayloader, *rtpsession, *sink; GstBus *bus = NULL; gboolean res = FALSE; GstSegment segment; GstPad *sinkpad = NULL; GstPad *rtcp_sinkpad = NULL; GstPad *fake_udp_sinkpad = NULL; GstPad *rtcp_srcpad = NULL; GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE; GST_INFO ("preparing test"); nb_ssrc_changes = 0; ssrc_prev = -1; /* 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", 5, NULL); encoder = gst_element_factory_make ("speexenc", NULL); rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); g_object_set (rtppayloader, "pt", 96, NULL); rtpsession = gst_element_factory_make ("rtpsession", NULL); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader, rtpsession, sink, NULL); /* link elements */ res = gst_element_link (src, encoder); fail_unless (res == TRUE, NULL); res = gst_element_link (encoder, rtppayloader); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtppayloader, "src", rtpsession, "send_rtp_sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpsession, "send_rtp_src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); /* add probe on rtpsession sink pad to induce collision */ sinkpad = gst_element_get_static_pad (rtpsession, "send_rtp_sink"); gst_pad_add_probe (sinkpad, (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH), (GstPadProbeCallback) rtpsession_sinkpad_probe, NULL, NULL); gst_object_unref (sinkpad); /* setup rtcp link */ srcpad = gst_pad_new_from_static_template (&srctemplate, "src"); rtcp_sinkpad = gst_element_get_request_pad (rtpsession, "recv_rtcp_sink"); fail_unless (gst_pad_link (srcpad, rtcp_sinkpad) == GST_PAD_LINK_OK, NULL); gst_object_unref (rtcp_sinkpad); res = gst_pad_set_active (srcpad, TRUE); fail_if (res == FALSE); res = gst_pad_push_event (srcpad, gst_event_new_stream_start ("my_rtcp_stream_id")); fail_if (res == FALSE); gst_segment_init (&segment, GST_FORMAT_TIME); res = gst_pad_push_event (srcpad, gst_event_new_segment (&segment)); fail_if (res == FALSE); fake_udp_sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); gst_pad_set_chain_function (fake_udp_sinkpad, fake_udp_sink_chain_func); rtcp_srcpad = gst_element_get_request_pad (rtpsession, "send_rtcp_src"); fail_unless (gst_pad_link (rtcp_srcpad, fake_udp_sinkpad) == GST_PAD_LINK_OK, NULL); gst_object_unref (rtcp_srcpad); res = gst_pad_set_active (fake_udp_sinkpad, TRUE); fail_if (res == FALSE); /* connect messages */ 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) message_received, bin); state_res = gst_element_set_state (bin, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); 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); /* cleanup */ gst_object_unref (srcpad); gst_object_unref (fake_udp_sinkpad); g_main_loop_unref (main_loop); gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (bin); /* check results */ fail_unless_equals_int (nb_ssrc_changes, 7); } GST_END_TEST; static guint ssrc_before; static guint ssrc_after; static guint rtx_ssrc_before; static guint rtx_ssrc_after; static GstPadProbeReturn rtpsession_sinkpad_probe2 (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstPadProbeReturn ret = GST_PAD_PROBE_OK; if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) { GstBuffer *buffer = GST_BUFFER (info->data); GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; guint payload_type = 0; static gint i = 0; /* retrieve current ssrc for retransmission stream only */ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp); payload_type = gst_rtp_buffer_get_payload_type (&rtp); if (payload_type == 99) { if (i < 3) rtx_ssrc_before = gst_rtp_buffer_get_ssrc (&rtp); else rtx_ssrc_after = gst_rtp_buffer_get_ssrc (&rtp); } else { /* ask to retransmit every packet */ GstEvent *event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, gst_structure_new ("GstRTPRetransmissionRequest", "seqnum", G_TYPE_UINT, gst_rtp_buffer_get_seq (&rtp), "ssrc", G_TYPE_UINT, gst_rtp_buffer_get_ssrc (&rtp), NULL)); gst_pad_push_event (pad, event); if (i < 3) ssrc_before = gst_rtp_buffer_get_ssrc (&rtp); else ssrc_after = gst_rtp_buffer_get_ssrc (&rtp); } gst_rtp_buffer_unmap (&rtp); /* feint a collision on recv_rtcp_sink pad of gstrtpsession * (note that after being marked as collied the rtpsession ignores * all non bye packets) */ if (i == 2) { GstBuffer *rtcp_buffer = create_rtcp_app (rtx_ssrc_before); /* push collied packet on recv_rtcp_sink */ gst_pad_push (srcpad, rtcp_buffer); } ++i; } return ret; } /* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ * rtprtxsend ! rtpsession ! fakesink * It manually pushs buffer into rtpsession with same ssrc than rtx stream * but different ip so that collision can be detected * The test checks that the rtx elements changes its ssrc whereas * the payloader keeps its master ssrc */ GST_START_TEST (test_rtx_ssrc_collision) { GstElement *bin, *src, *encoder, *rtppayloader, *rtprtxsend, *rtpsession, *sink; GstBus *bus = NULL; gboolean res = FALSE; GstSegment segment; GstPad *sinkpad = NULL; GstPad *rtcp_sinkpad = NULL; GstPad *fake_udp_sinkpad = NULL; GstPad *rtcp_srcpad = NULL; GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE; GstStructure *pt_map; 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", 5, NULL); encoder = gst_element_factory_make ("speexenc", NULL); rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); g_object_set (rtppayloader, "pt", 96, NULL); rtprtxsend = gst_element_factory_make ("rtprtxsend", NULL); pt_map = gst_structure_new ("application/x-rtp-pt-map", "96", G_TYPE_UINT, 99, NULL); g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL); gst_structure_free (pt_map); rtpsession = gst_element_factory_make ("rtpsession", NULL); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader, rtprtxsend, rtpsession, sink, NULL); /* link elements */ res = gst_element_link (src, encoder); fail_unless (res == TRUE, NULL); res = gst_element_link (encoder, rtppayloader); fail_unless (res == TRUE, NULL); res = gst_element_link (rtppayloader, rtprtxsend); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtprtxsend, "src", rtpsession, "send_rtp_sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpsession, "send_rtp_src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); /* add probe on rtpsession sink pad to induce collision */ sinkpad = gst_element_get_static_pad (rtpsession, "send_rtp_sink"); gst_pad_add_probe (sinkpad, (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH), (GstPadProbeCallback) rtpsession_sinkpad_probe2, NULL, NULL); gst_object_unref (sinkpad); /* setup rtcp link */ srcpad = gst_pad_new_from_static_template (&srctemplate, "src"); rtcp_sinkpad = gst_element_get_request_pad (rtpsession, "recv_rtcp_sink"); fail_unless (gst_pad_link (srcpad, rtcp_sinkpad) == GST_PAD_LINK_OK, NULL); gst_object_unref (rtcp_sinkpad); res = gst_pad_set_active (srcpad, TRUE); fail_if (res == FALSE); res = gst_pad_push_event (srcpad, gst_event_new_stream_start ("my_rtcp_stream_id")); fail_if (res == FALSE); gst_segment_init (&segment, GST_FORMAT_TIME); res = gst_pad_push_event (srcpad, gst_event_new_segment (&segment)); fail_if (res == FALSE); fake_udp_sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); gst_pad_set_chain_function (fake_udp_sinkpad, fake_udp_sink_chain_func); rtcp_srcpad = gst_element_get_request_pad (rtpsession, "send_rtcp_src"); fail_unless (gst_pad_link (rtcp_srcpad, fake_udp_sinkpad) == GST_PAD_LINK_OK, NULL); gst_object_unref (rtcp_srcpad); res = gst_pad_set_active (fake_udp_sinkpad, TRUE); fail_if (res == FALSE); /* connect messages */ 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) message_received, bin); state_res = gst_element_set_state (bin, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); 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); /* cleanup */ gst_object_unref (srcpad); gst_object_unref (fake_udp_sinkpad); g_main_loop_unref (main_loop); gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (bin); /* check results */ fail_if (rtx_ssrc_before == rtx_ssrc_after); fail_if (ssrc_before != ssrc_after); } GST_END_TEST; static Suite * rtpcollision_suite (void) { Suite *s = suite_create ("rtpcollision"); TCase *tc_chain = tcase_create ("general"); tcase_set_timeout (tc_chain, 10); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_master_ssrc_collision); tcase_add_test (tc_chain, test_rtx_ssrc_collision); return s; } GST_CHECK_MAIN (rtpcollision);