gstreamer/tests/examples/rtp/client-rtpbundle.c

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/* 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>
/*
* 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;
}