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dcd3ce9751
A new signal named on-bundled-ssrc is provided and can be used by the application to redirect a stream to a different GstRtpSession or to keep the RTX stream grouped within the GstRtpSession of the same media type. https://bugzilla.gnome.org/show_bug.cgi?id=772740
179 lines
7.4 KiB
C
179 lines
7.4 KiB
C
/* GStreamer
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* Copyright (C) 2016 Igalia S.L
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* @author Philippe Normand <philn@igalia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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/*
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* An bundling RTP server
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* creates two sessions and streams audio on one, video on the other, with RTCP
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* on both sessions. The destination is 127.0.0.1.
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*
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* The RTP streams are bundled to a single outgoing connection. Same for the RTCP streams.
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*
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* .-------. .-------. .-------. .------------. .------.
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* |audiots| |alawenc| |pcmapay| | rtpbin | |funnel|
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* | src->sink src->sink src->send_rtp_0 send_rtp_0--->sink_0 | .-------.
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* '-------' '-------' '-------' | | | | |udpsink|
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* | | | src->sink |
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* .-------. .---------. | | | | '-------'
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* |videots| | vrawpay | | | | |
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* | src------------>sink src->send_rtp_1 send_rtp_1--->sink_1 |
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* '-------' '---------' | | '------'
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* | |
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* .------. | |
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* |udpsrc| | | .------.
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* | src->recv_rtcp_0 | |funnel|
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* '------' | send_rtcp_0-->sink_0 | .-------.
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* | | | | |udpsink|
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* .------. | | | src->sink |
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* |udpsrc| | | | | '-------'
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* | src->recv_rtcp_1 | | |
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* '------' | send_rtcp_1-->sink_1 |
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* '------------' '------'
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*
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*/
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static GstElement *
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create_pipeline (void)
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{
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GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
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*audio_rtppayloader, *sendrtp_udpsink,
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*send_rtcp_udpsink, *sendrtcp_funnel, *sendrtp_funnel;
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GstElement *videosrc, *video_rtppayloader, *time_overlay;
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gint rtp_udp_port = 5001;
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gint rtcp_udp_port = 5002;
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gint recv_audio_rtcp_port = 5003;
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gint recv_video_rtcp_port = 5004;
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GstElement *audio_rtcp_udpsrc, *video_rtcp_udpsrc;
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pipeline = gst_pipeline_new (NULL);
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rtpbin = gst_element_factory_make ("rtpbin", NULL);
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audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
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g_object_set (audiosrc, "is-live", TRUE, NULL);
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audio_encoder = gst_element_factory_make ("alawenc", NULL);
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audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
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g_object_set (audio_rtppayloader, "pt", 96, NULL);
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videosrc = gst_element_factory_make ("videotestsrc", NULL);
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g_object_set (videosrc, "is-live", TRUE, NULL);
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time_overlay = gst_element_factory_make ("timeoverlay", NULL);
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video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
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g_object_set (video_rtppayloader, "pt", 100, NULL);
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/* muxed rtcp */
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sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
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send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
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g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
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g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
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g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
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g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
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/* outgoing bundled stream */
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sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
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sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
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g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
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g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
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g_object_set (sendrtp_udpsink, "sync", FALSE, NULL);
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g_object_set (sendrtp_udpsink, "async", FALSE, NULL);
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gst_bin_add_many (GST_BIN (pipeline), rtpbin, audiosrc, audio_encoder,
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audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
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sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
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if (time_overlay)
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gst_bin_add (GST_BIN (pipeline), time_overlay);
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gst_element_link_many (audiosrc, audio_encoder, audio_rtppayloader, NULL);
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gst_element_link_pads (audio_rtppayloader, "src", rtpbin, "send_rtp_sink_0");
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if (time_overlay) {
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gst_element_link_many (videosrc, time_overlay, video_rtppayloader, NULL);
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} else {
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gst_element_link (videosrc, video_rtppayloader);
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}
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gst_element_link_pads (video_rtppayloader, "src", rtpbin, "send_rtp_sink_1");
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gst_element_link_pads (sendrtp_funnel, "src", sendrtp_udpsink, "sink");
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gst_element_link_pads (rtpbin, "send_rtp_src_0", sendrtp_funnel, "sink_%u");
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gst_element_link_pads (rtpbin, "send_rtp_src_1", sendrtp_funnel, "sink_%u");
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gst_element_link_pads (sendrtcp_funnel, "src", send_rtcp_udpsink, "sink");
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gst_element_link_pads (rtpbin, "send_rtcp_src_0", sendrtcp_funnel, "sink_%u");
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gst_element_link_pads (rtpbin, "send_rtcp_src_1", sendrtcp_funnel, "sink_%u");
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audio_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
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g_object_set (audio_rtcp_udpsrc, "port", recv_audio_rtcp_port, NULL);
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video_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
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g_object_set (video_rtcp_udpsrc, "port", recv_video_rtcp_port, NULL);
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gst_bin_add_many (GST_BIN (pipeline), audio_rtcp_udpsrc, video_rtcp_udpsrc,
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NULL);
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gst_element_link_pads (audio_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0");
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gst_element_link_pads (video_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_1");
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return pipeline;
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}
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/*
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* Used to generate informative messages during pipeline startup
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*/
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static void
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cb_state (GstBus * bus, GstMessage * message, gpointer data)
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{
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GstObject *pipe = GST_OBJECT (data);
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GstState old, new, pending;
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gst_message_parse_state_changed (message, &old, &new, &pending);
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if (message->src == pipe) {
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g_print ("Pipeline %s changed state from %s to %s\n",
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GST_OBJECT_NAME (message->src),
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gst_element_state_get_name (old), gst_element_state_get_name (new));
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}
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}
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int
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main (int argc, char **argv)
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{
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GstElement *pipe;
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GstBus *bus;
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GMainLoop *loop;
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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pipe = create_pipeline ();
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bus = gst_element_get_bus (pipe);
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g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
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gst_bus_add_signal_watch (bus);
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gst_object_unref (bus);
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g_print ("starting server pipeline\n");
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gst_element_set_state (pipe, GST_STATE_PLAYING);
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g_main_loop_run (loop);
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g_print ("stopping server pipeline\n");
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gst_element_set_state (pipe, GST_STATE_NULL);
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gst_object_unref (pipe);
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g_main_loop_unref (loop);
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return 0;
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}
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