/* 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 gboolean send_pipeline_eos = FALSE; static gboolean receive_pipeline_eos = 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: if (!strcmp ("pipeline_send", GST_OBJECT_NAME (GST_MESSAGE_SRC (message)))) send_pipeline_eos = TRUE; else if (!strcmp ("pipeline_receive", GST_OBJECT_NAME (GST_MESSAGE_SRC (message)))) receive_pipeline_eos = TRUE; else fail ("Unknown pipeline: %s", GST_OBJECT_NAME (GST_MESSAGE_SRC (message))); 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; } } typedef struct { guint count; guint nb_packets; guint drop_every_n_packets; } RTXSendData; static GstPadProbeReturn rtprtxsend_srcpad_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); RTXSendData *rtxdata = (RTXSendData *) user_data; GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; guint payload_type = 0; gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp); payload_type = gst_rtp_buffer_get_payload_type (&rtp); /* main stream packets */ if (payload_type == 96) { /* count packets of the main stream */ ++rtxdata->nb_packets; /* drop some packets */ if (rtxdata->count < rtxdata->drop_every_n_packets) { ++rtxdata->count; } else { /* drop a packet every 'rtxdata->count' packets */ rtxdata->count = 1; ret = GST_PAD_PROBE_DROP; } } else { /* retransmission packets */ } gst_rtp_buffer_unmap (&rtp); } return ret; } static void on_rtpbinreceive_pad_added (GstElement * element, GstPad * newPad, gpointer data) { GstElement *rtpdepayloader = GST_ELEMENT (data); gchar *padName = gst_pad_get_name (newPad); if (g_str_has_prefix (padName, "recv_rtp_src_")) { GstPad *sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink"); gst_pad_link (newPad, sinkpad); gst_object_unref (sinkpad); } g_free (padName); } static gboolean on_timeout (gpointer data) { GstEvent *eos = gst_event_new_eos (); if (!gst_element_send_event (GST_ELEMENT (data), eos)) { GST_ERROR ("failed to send end of stream event"); gst_event_unref (eos); } return FALSE; } static GstElement * request_aux_receive (GstElement * rtpbin, guint sessid, GstElement * receive) { GstElement *bin; GstPad *pad; GST_INFO ("creating AUX receiver"); bin = gst_bin_new (NULL); gst_bin_add (GST_BIN (bin), receive); pad = gst_element_get_static_pad (receive, "src"); gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad)); gst_object_unref (pad); pad = gst_element_get_static_pad (receive, "sink"); gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad)); gst_object_unref (pad); return bin; } static GstElement * request_aux_send (GstElement * rtpbin, guint sessid, GstElement * send) { GstElement *bin; GstPad *pad; GST_INFO ("creating AUX sender"); bin = gst_bin_new (NULL); gst_bin_add (GST_BIN (bin), send); pad = gst_element_get_static_pad (send, "src"); gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad)); gst_object_unref (pad); pad = gst_element_get_static_pad (send, "sink"); gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad)); gst_object_unref (pad); return bin; } GST_START_TEST (test_simple_rtpbin_aux) { GstElement *binsend, *rtpbinsend, *src, *encoder, *rtppayloader, *rtprtxsend, *sendrtp_udpsink, *sendrtcp_udpsink, *sendrtcp_udpsrc; GstElement *binreceive, *rtpbinreceive, *recvrtp_udpsrc, *recvrtcp_udpsrc, *recvrtcp_udpsink, *rtprtxreceive, *rtpdepayloader, *decoder, *converter, *sink; GstBus *bussend; GstBus *busreceive; gboolean res; GstCaps *rtpcaps = NULL; GstStructure *pt_map; GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE; GstPad *srcpad = NULL; guint nb_rtx_send_packets = 0; guint nb_rtx_recv_packets = 0; RTXSendData send_rtxdata; send_rtxdata.count = 1; send_rtxdata.nb_packets = 0; send_rtxdata.drop_every_n_packets = 25; GST_INFO ("preparing test"); /* build pipeline */ binsend = gst_pipeline_new ("pipeline_send"); bussend = gst_element_get_bus (binsend); gst_bus_add_signal_watch_full (bussend, G_PRIORITY_HIGH); binreceive = gst_pipeline_new ("pipeline_receive"); busreceive = gst_element_get_bus (binreceive); gst_bus_add_signal_watch_full (busreceive, G_PRIORITY_HIGH); rtpbinsend = gst_element_factory_make ("rtpbin", "rtpbinsend"); g_object_set (rtpbinsend, "latency", 200, "do-retransmission", TRUE, NULL); src = gst_element_factory_make ("audiotestsrc", "src"); encoder = gst_element_factory_make ("alawenc", "encoder"); rtppayloader = gst_element_factory_make ("rtppcmapay", "rtppayloader"); rtprtxsend = gst_element_factory_make ("rtprtxsend", "rtprtxsend"); sendrtp_udpsink = gst_element_factory_make ("udpsink", "sendrtp_udpsink"); g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL); g_object_set (sendrtp_udpsink, "port", 5006, NULL); sendrtcp_udpsink = gst_element_factory_make ("udpsink", "sendrtcp_udpsink"); g_object_set (sendrtcp_udpsink, "host", "127.0.0.1", NULL); g_object_set (sendrtcp_udpsink, "port", 5007, NULL); g_object_set (sendrtcp_udpsink, "sync", FALSE, NULL); g_object_set (sendrtcp_udpsink, "async", FALSE, NULL); sendrtcp_udpsrc = gst_element_factory_make ("udpsrc", "sendrtcp_udpsrc"); g_object_set (sendrtcp_udpsrc, "port", 5009, NULL); rtpbinreceive = gst_element_factory_make ("rtpbin", "rtpbinreceive"); g_object_set (rtpbinreceive, "latency", 200, "do-retransmission", TRUE, NULL); recvrtp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtp_udpsrc"); g_object_set (recvrtp_udpsrc, "port", 5006, NULL); rtpcaps = gst_caps_from_string ("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)96"); g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL); gst_caps_unref (rtpcaps); recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc"); g_object_set (recvrtcp_udpsrc, "port", 5007, NULL); recvrtcp_udpsink = gst_element_factory_make ("udpsink", "recvrtcp_udpsink"); g_object_set (recvrtcp_udpsink, "host", "127.0.0.1", NULL); g_object_set (recvrtcp_udpsink, "port", 5009, NULL); g_object_set (recvrtcp_udpsink, "sync", FALSE, NULL); g_object_set (recvrtcp_udpsink, "async", FALSE, NULL); rtprtxreceive = gst_element_factory_make ("rtprtxreceive", "rtprtxreceive"); rtpdepayloader = gst_element_factory_make ("rtppcmadepay", "rtpdepayloader"); decoder = gst_element_factory_make ("alawdec", "decoder"); converter = gst_element_factory_make ("identity", "converter"); sink = gst_element_factory_make ("fakesink", "sink"); g_object_set (sink, "sync", TRUE, NULL); gst_bin_add_many (GST_BIN (binsend), rtpbinsend, src, encoder, rtppayloader, sendrtp_udpsink, sendrtcp_udpsink, sendrtcp_udpsrc, NULL); gst_bin_add_many (GST_BIN (binreceive), rtpbinreceive, recvrtp_udpsrc, recvrtcp_udpsrc, recvrtcp_udpsink, rtpdepayloader, decoder, converter, sink, NULL); g_signal_connect (rtpbinreceive, "pad-added", G_CALLBACK (on_rtpbinreceive_pad_added), rtpdepayloader); pt_map = gst_structure_new ("application/x-rtp-pt-map", "96", G_TYPE_UINT, 99, NULL); g_object_set (rtppayloader, "pt", 96, NULL); g_object_set (rtppayloader, "seqnum-offset", 1, NULL); g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL); g_object_set (rtprtxreceive, "payload-type-map", pt_map, NULL); gst_structure_free (pt_map); /* set rtp aux receive */ g_signal_connect (rtpbinreceive, "request-aux-receiver", (GCallback) request_aux_receive, rtprtxreceive); /* set rtp aux send */ g_signal_connect (rtpbinsend, "request-aux-sender", (GCallback) request_aux_send, rtprtxsend); /* gst-launch-1.0 rtpbin name=rtpbin audiotestsrc ! amrnbenc ! rtpamrpay ! \ * rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host="127.0.0.1" \ * port=5002 rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5003 \ * sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 */ 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", rtpbinsend, "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpbinsend, "send_rtp_src_0", sendrtp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpbinsend, "send_rtcp_src_0", sendrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (sendrtcp_udpsrc, "src", rtpbinsend, "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); srcpad = gst_element_get_static_pad (rtpbinsend, "send_rtp_src_0"); gst_pad_add_probe (srcpad, (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH), (GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL); gst_object_unref (srcpad); /* gst-launch-1.0 rtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio, \ * clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,o * ctet-align=(string)1" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! \ * amrnbdec ! fakesink sync=True udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ * rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5007 sync=false async=false */ res = gst_element_link_pads_full (recvrtp_udpsrc, "src", rtpbinreceive, "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpdepayloader, "src", decoder, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link (decoder, converter); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (converter, "src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (recvrtcp_udpsrc, "src", rtpbinreceive, "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); res = gst_element_link_pads_full (rtpbinreceive, "send_rtcp_src_0", recvrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING); fail_unless (res == TRUE, NULL); g_signal_connect (bussend, "message::error", (GCallback) message_received, binsend); g_signal_connect (bussend, "message::warning", (GCallback) message_received, binsend); g_signal_connect (bussend, "message::eos", (GCallback) message_received, binsend); g_signal_connect (busreceive, "message::error", (GCallback) message_received, binreceive); g_signal_connect (busreceive, "message::warning", (GCallback) message_received, binreceive); g_signal_connect (busreceive, "message::eos", (GCallback) message_received, binreceive); state_res = gst_element_set_state (binreceive, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); state_res = gst_element_set_state (binsend, GST_STATE_PLAYING); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); g_timeout_add (5000, on_timeout, binsend); g_timeout_add (5000, on_timeout, binreceive); GST_INFO ("enter mainloop"); while (!send_pipeline_eos && !receive_pipeline_eos) g_main_context_iteration (NULL, TRUE); GST_INFO ("exit mainloop"); /* check that FB NACK is working */ g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nb_rtx_send_packets, NULL); g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests", &nb_rtx_recv_packets, NULL); state_res = gst_element_set_state (binsend, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); state_res = gst_element_set_state (binreceive, GST_STATE_NULL); ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); GST_INFO ("nb_rtx_send_packets %d", nb_rtx_send_packets); GST_INFO ("nb_rtx_recv_packets %d", nb_rtx_recv_packets); fail_if (nb_rtx_send_packets < 1); fail_if (nb_rtx_recv_packets < 1); /* cleanup */ gst_bus_remove_signal_watch (bussend); gst_object_unref (bussend); gst_object_unref (binsend); gst_bus_remove_signal_watch (busreceive); gst_object_unref (busreceive); gst_object_unref (binreceive); } GST_END_TEST; static Suite * rtpaux_suite (void) { Suite *s = suite_create ("rtpaux"); 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_aux); return s; } GST_CHECK_MAIN (rtpaux);