| /* GStreamer |
| * Copyright (C) 2016 Igalia S.L |
| * @author Philippe Normand <philn@igalia.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. |
| */ |
| #include <gst/gst.h> |
| |
| /* |
| * An bundling RTP server |
| * creates two sessions and streams audio on one, video on the other, with RTCP |
| * on both sessions. The destination is 127.0.0.1. |
| * |
| * The RTP streams are bundled to a single outgoing connection. Same for the RTCP streams. |
| * |
| * .-------. .-------. .-------. .------------. .------. |
| * |audiots| |alawenc| |pcmapay| | rtpbin | |funnel| |
| * | src->sink src->sink src->send_rtp_0 send_rtp_0--->sink_0 | .-------. |
| * '-------' '-------' '-------' | | | | |udpsink| |
| * | | | src->sink | |
| * .-------. .---------. | | | | '-------' |
| * |videots| | vrawpay | | | | | |
| * | src------------>sink src->send_rtp_1 send_rtp_1--->sink_1 | |
| * '-------' '---------' | | '------' |
| * | | |
| * .------. | | |
| * |udpsrc| | | .------. |
| * | src->recv_rtcp_0 | |funnel| |
| * '------' | send_rtcp_0-->sink_0 | .-------. |
| * | | | | |udpsink| |
| * .------. | | | src->sink | |
| * |udpsrc| | | | | '-------' |
| * | src->recv_rtcp_1 | | | |
| * '------' | send_rtcp_1-->sink_1 | |
| * '------------' '------' |
| * |
| */ |
| |
| static GstElement * |
| create_pipeline (void) |
| { |
| GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder, |
| *audio_rtppayloader, *sendrtp_udpsink, |
| *send_rtcp_udpsink, *sendrtcp_funnel, *sendrtp_funnel; |
| GstElement *videosrc, *video_rtppayloader, *time_overlay; |
| gint rtp_udp_port = 5001; |
| gint rtcp_udp_port = 5002; |
| gint recv_audio_rtcp_port = 5003; |
| gint recv_video_rtcp_port = 5004; |
| GstElement *audio_rtcp_udpsrc, *video_rtcp_udpsrc; |
| |
| pipeline = gst_pipeline_new (NULL); |
| |
| rtpbin = gst_element_factory_make ("rtpbin", NULL); |
| |
| audiosrc = gst_element_factory_make ("audiotestsrc", NULL); |
| g_object_set (audiosrc, "is-live", TRUE, 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); |
| |
| videosrc = gst_element_factory_make ("videotestsrc", NULL); |
| g_object_set (videosrc, "is-live", TRUE, NULL); |
| time_overlay = gst_element_factory_make ("timeoverlay", NULL); |
| video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL); |
| g_object_set (video_rtppayloader, "pt", 100, 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); |
| g_object_set (sendrtp_udpsink, "sync", FALSE, NULL); |
| g_object_set (sendrtp_udpsink, "async", FALSE, NULL); |
| |
| gst_bin_add_many (GST_BIN (pipeline), rtpbin, audiosrc, audio_encoder, |
| audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink, |
| sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL); |
| |
| if (time_overlay) |
| gst_bin_add (GST_BIN (pipeline), time_overlay); |
| |
| gst_element_link_many (audiosrc, audio_encoder, audio_rtppayloader, NULL); |
| gst_element_link_pads (audio_rtppayloader, "src", rtpbin, "send_rtp_sink_0"); |
| |
| if (time_overlay) { |
| gst_element_link_many (videosrc, time_overlay, video_rtppayloader, NULL); |
| } else { |
| gst_element_link (videosrc, video_rtppayloader); |
| } |
| |
| gst_element_link_pads (video_rtppayloader, "src", rtpbin, "send_rtp_sink_1"); |
| |
| gst_element_link_pads (sendrtp_funnel, "src", sendrtp_udpsink, "sink"); |
| gst_element_link_pads (rtpbin, "send_rtp_src_0", sendrtp_funnel, "sink_%u"); |
| gst_element_link_pads (rtpbin, "send_rtp_src_1", sendrtp_funnel, "sink_%u"); |
| gst_element_link_pads (sendrtcp_funnel, "src", send_rtcp_udpsink, "sink"); |
| gst_element_link_pads (rtpbin, "send_rtcp_src_0", sendrtcp_funnel, "sink_%u"); |
| gst_element_link_pads (rtpbin, "send_rtcp_src_1", sendrtcp_funnel, "sink_%u"); |
| |
| audio_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL); |
| g_object_set (audio_rtcp_udpsrc, "port", recv_audio_rtcp_port, NULL); |
| video_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL); |
| g_object_set (video_rtcp_udpsrc, "port", recv_video_rtcp_port, NULL); |
| gst_bin_add_many (GST_BIN (pipeline), audio_rtcp_udpsrc, video_rtcp_udpsrc, |
| NULL); |
| gst_element_link_pads (audio_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0"); |
| gst_element_link_pads (video_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_1"); |
| |
| 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)); |
| } |
| } |
| |
| 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; |
| } |
