/* 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 /* * RTP bundle receiver * * In this example we initially create one RTP session but the incoming RTP * and RTCP streams actually bundle 2 different media type, one audio stream * and one video stream. We are notified of the discovery of the streams by * the on-bundled-ssrc rtpbin signal. In the handler we decide to assign the * first SSRC to the (existing) audio session and the second SSRC to a new * session (id: 1). * * .-------. .----------. .-----------. .-------. .-------------. * RTP |udpsrc | | rtpbin | | pcmadepay | |alawdec| |autoaudiosink| * port=5001 | src->recv_rtp_0 recv_rtp_0->sink src->sink src->sink | * '-------' | | '-----------' '-------' '-------------' * | | * | | .-------. * | | |udpsink| RTCP * | send_rtcp_0->sink | port=5003 * .-------. | | '-------' sync=false * RTCP |udpsrc | | | async=false * port=5002 | src->recv_rtcp_0 | * '-------' | | * | | * | | .---------. .-------------. * | | |vrawdepay| |autovideosink| * | recv_rtp_1->sink src->sink | * | | '---------' '-------------' * | | * | | .-------. * | | |udpsink| RTCP * | send_rtcp_1->sink | port=5004 * | | '-------' sync=false * | | async=false * | | * '----------' * */ static gboolean plug_video_rtcp_sender (gpointer user_data) { gint send_video_rtcp_port = 5004; GstElement *rtpbin = GST_ELEMENT_CAST (user_data); GstElement *send_video_rtcp_udpsink; GstElement *pipeline = GST_ELEMENT_CAST (gst_object_get_parent (GST_OBJECT (rtpbin))); send_video_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL); g_object_set (send_video_rtcp_udpsink, "host", "127.0.0.1", NULL); g_object_set (send_video_rtcp_udpsink, "port", send_video_rtcp_port, NULL); g_object_set (send_video_rtcp_udpsink, "sync", FALSE, NULL); g_object_set (send_video_rtcp_udpsink, "async", FALSE, NULL); gst_bin_add (GST_BIN (pipeline), send_video_rtcp_udpsink); gst_element_link_pads (rtpbin, "send_rtcp_src_1", send_video_rtcp_udpsink, "sink"); gst_element_sync_state_with_parent (send_video_rtcp_udpsink); gst_object_unref (pipeline); gst_object_unref (rtpbin); return G_SOURCE_REMOVE; } static void on_rtpbinreceive_pad_added (GstElement * rtpbin, 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); 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); if (g_str_has_prefix (pad_name, "recv_rtp_src_1")) { g_timeout_add (0, plug_video_rtcp_sender, gst_object_ref (rtpbin)); } } 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 (void) { GstElement *pipeline, *rtpbin, *recv_rtp_udpsrc, *recv_rtcp_udpsrc, *audio_rtpdepayloader, *audio_decoder, *audio_sink, *video_rtpdepayloader, *video_sink, *send_audio_rtcp_udpsink; GstCaps *rtpcaps; gint rtp_udp_port = 5001; gint rtcp_udp_port = 5002; gint send_audio_rtcp_port = 5003; pipeline = gst_pipeline_new (NULL); rtpbin = gst_element_factory_make ("rtpbin", NULL); g_object_set (rtpbin, "latency", 200, NULL); 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); 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 ("autoaudiosink", NULL); video_rtpdepayloader = gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader"); video_sink = gst_element_factory_make ("autovideosink", 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); gst_element_link_pads (audio_rtpdepayloader, "src", audio_decoder, "sink"); gst_element_link (audio_decoder, audio_sink); gst_element_link_pads (video_rtpdepayloader, "src", video_sink, "sink"); /* request a single receiving RTP session. */ gst_element_link_pads (recv_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0"); gst_element_link_pads (recv_rtp_udpsrc, "src", rtpbin, "recv_rtp_sink_0"); send_audio_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL); g_object_set (send_audio_rtcp_udpsink, "host", "127.0.0.1", NULL); g_object_set (send_audio_rtcp_udpsink, "port", send_audio_rtcp_port, NULL); g_object_set (send_audio_rtcp_udpsink, "sync", FALSE, NULL); g_object_set (send_audio_rtcp_udpsink, "async", FALSE, NULL); gst_bin_add (GST_BIN (pipeline), send_audio_rtcp_udpsink); gst_element_link_pads (rtpbin, "send_rtcp_src_0", send_audio_rtcp_udpsink, "sink"); return pipeline; } /* * Used to generate informative messages during pipeline startup */ static void cb_state (GstBus * bus, GstMessage * message, gpointer data) { GstObject *pipe = GST_OBJECT (data); GstState old, new, pending; gst_message_parse_state_changed (message, &old, &new, &pending); if (message->src == pipe) { g_print ("Pipeline %s changed state from %s to %s\n", GST_OBJECT_NAME (message->src), gst_element_state_get_name (old), gst_element_state_get_name (new)); if (old == GST_STATE_PAUSED && new == GST_STATE_PLAYING) GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipe), GST_DEBUG_GRAPH_SHOW_ALL, GST_OBJECT_NAME (message->src)); } } int main (int argc, char **argv) { GstElement *pipe; GstBus *bus; GMainLoop *loop; gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); pipe = create_pipeline (); bus = gst_element_get_bus (pipe); g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe); gst_bus_add_signal_watch (bus); gst_object_unref (bus); g_print ("starting server pipeline\n"); gst_element_set_state (pipe, GST_STATE_PLAYING); g_main_loop_run (loop); g_print ("stopping server pipeline\n"); gst_element_set_state (pipe, GST_STATE_NULL); gst_object_unref (pipe); g_main_loop_unref (loop); return 0; }