/* GStreamer
* Copyright (C) 2013 Collabora Ltd.
* @author Torrie Fischer <torrie.fischer@collabora.co.uk>
*
* 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>
#include <gst/rtp/rtp.h>
#include <stdlib.h>
/*
* RTP receiver with RFC4588 retransmission handling enabled
*
* In this example we have two RTP sessions, one for video and one for audio.
* Video is received on port 5000, with its RTCP stream received on port 5001
* and sent on port 5005. Audio is received on port 5005, with its RTCP stream
* received on port 5006 and sent on port 5011.
*
* In both sessions, we set "rtprtxreceive" as the session's "aux" element
* in rtpbin, which enables RFC4588 retransmission handling for that session.
*
* .-------. .----------. .-----------. .---------. .-------------.
* RTP |udpsrc | | rtpbin | |theoradepay| |theoradec| |autovideosink|
* port=5000 | src->recv_rtp_0 recv_rtp_0->sink src->sink src->sink |
* '-------' | | '-----------' '---------' '-------------'
* | |
* | | .-------.
* | | |udpsink| RTCP
* | send_rtcp_0->sink | port=5005
* .-------. | | '-------' sync=false
* RTCP |udpsrc | | | async=false
* port=5001 | src->recv_rtcp_0 |
* '-------' | |
* | |
* .-------. | | .---------. .-------. .-------------.
* RTP |udpsrc | | | |pcmadepay| |alawdec| |autoaudiosink|
* port=5006 | src->recv_rtp_1 recv_rtp_1->sink src->sink src->sink |
* '-------' | | '---------' '-------' '-------------'
* | |
* | | .-------.
* | | |udpsink| RTCP
* | send_rtcp_1->sink | port=5011
* .-------. | | '-------' sync=false
* RTCP |udpsrc | | | async=false
* port=5007 | src->recv_rtcp_1 |
* '-------' '----------'
*
*/
GMainLoop *loop = NULL;
typedef struct _SessionData
{
int ref;
GstElement *rtpbin;
guint sessionNum;
GstCaps *caps;
GstElement *output;
} SessionData;
static SessionData *
session_ref (SessionData * data)
{
g_atomic_int_inc (&data->ref);
return data;
}
static void
session_unref (gpointer data)
{
SessionData *session = (SessionData *) data;
if (g_atomic_int_dec_and_test (&session->ref)) {
g_object_unref (session->rtpbin);
gst_caps_unref (session->caps);
g_free (session);
}
}
static SessionData *
session_new (guint sessionNum)
{
SessionData *ret = g_new0 (SessionData, 1);
ret->sessionNum = sessionNum;
return session_ref (ret);
}
static void
setup_ghost_sink (GstElement * sink, GstBin * bin)
{
GstPad *sinkPad = gst_element_get_static_pad (sink, "sink");
GstPad *binPad = gst_ghost_pad_new ("sink", sinkPad);
gst_element_add_pad (GST_ELEMENT (bin), binPad);
}
static SessionData *
make_audio_session (guint sessionNum)
{
SessionData *ret = session_new (sessionNum);
GstBin *bin = GST_BIN (gst_bin_new ("audio"));
GstElement *queue = gst_element_factory_make ("queue", NULL);
GstElement *sink = gst_element_factory_make ("autoaudiosink", NULL);
GstElement *audioconvert = gst_element_factory_make ("audioconvert", NULL);
GstElement *audioresample = gst_element_factory_make ("audioresample", NULL);
GstElement *depayloader = gst_element_factory_make ("rtppcmadepay", NULL);
GstElement *decoder = gst_element_factory_make ("alawdec", NULL);
gst_bin_add_many (bin, queue, depayloader, decoder, audioconvert,
audioresample, sink, NULL);
gst_element_link_many (queue, depayloader, decoder, audioconvert,
audioresample, sink, NULL);
setup_ghost_sink (queue, bin);
ret->output = GST_ELEMENT (bin);
ret->caps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"clock-rate", G_TYPE_INT, 8000,
"encoding-name", G_TYPE_STRING, "PCMA", NULL);
return ret;
}
static SessionData *
make_video_session (guint sessionNum)
{
SessionData *ret = session_new (sessionNum);
GstBin *bin = GST_BIN (gst_bin_new ("video"));
GstElement *queue = gst_element_factory_make ("queue", NULL);
GstElement *depayloader = gst_element_factory_make ("rtptheoradepay", NULL);
GstElement *decoder = gst_element_factory_make ("theoradec", NULL);
GstElement *converter = gst_element_factory_make ("videoconvert", NULL);
GstElement *sink = gst_element_factory_make ("autovideosink", NULL);
gst_bin_add_many (bin, depayloader, decoder, converter, queue, sink, NULL);
gst_element_link_many (queue, depayloader, decoder, converter, sink, NULL);
setup_ghost_sink (queue, bin);
ret->output = GST_ELEMENT (bin);
ret->caps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "video",
"clock-rate", G_TYPE_INT, 90000,
"encoding-name", G_TYPE_STRING, "THEORA", NULL);
return ret;
}
static GstCaps *
request_pt_map (GstElement * rtpbin, guint session, guint pt,
gpointer user_data)
{
SessionData *data = (SessionData *) user_data;
gchar *caps_str;
g_print ("Looking for caps for pt %u in session %u, have %u\n", pt, session,
data->sessionNum);
if (session == data->sessionNum) {
caps_str = gst_caps_to_string (data->caps);
g_print ("Returning %s\n", caps_str);
g_free (caps_str);
return gst_caps_ref (data->caps);
}
return NULL;
}
static void
cb_eos (GstBus * bus, GstMessage * message, gpointer data)
{
g_print ("Got EOS\n");
g_main_loop_quit (loop);
}
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));
}
}
static void
cb_warning (GstBus * bus, GstMessage * message, gpointer data)
{
GError *error = NULL;
gst_message_parse_warning (message, &error, NULL);
g_printerr ("Got warning from %s: %s\n", GST_OBJECT_NAME (message->src),
error->message);
g_error_free (error);
}
static void
cb_error (GstBus * bus, GstMessage * message, gpointer data)
{
GError *error = NULL;
gst_message_parse_error (message, &error, NULL);
g_printerr ("Got error from %s: %s\n", GST_OBJECT_NAME (message->src),
error->message);
g_error_free (error);
g_main_loop_quit (loop);
}
static void
handle_new_stream (GstElement * element, GstPad * newPad, gpointer data)
{
SessionData *session = (SessionData *) data;
gchar *padName;
gchar *myPrefix;
padName = gst_pad_get_name (newPad);
myPrefix = g_strdup_printf ("recv_rtp_src_%u", session->sessionNum);
g_print ("New pad: %s, looking for %s_*\n", padName, myPrefix);
if (g_str_has_prefix (padName, myPrefix)) {
GstPad *outputSinkPad;
GstElement *parent;
parent = GST_ELEMENT (gst_element_get_parent (session->rtpbin));
gst_bin_add (GST_BIN (parent), session->output);
gst_element_sync_state_with_parent (session->output);
gst_object_unref (parent);
outputSinkPad = gst_element_get_static_pad (session->output, "sink");
g_assert_cmpint (gst_pad_link (newPad, outputSinkPad), ==, GST_PAD_LINK_OK);
gst_object_unref (outputSinkPad);
g_print ("Linked!\n");
}
g_free (myPrefix);
g_free (padName);
}
static GstElement *
request_aux_receiver (GstElement * rtpbin, guint sessid, SessionData * session)
{
GstElement *rtx, *bin;
GstPad *pad;
gchar *name;
GstStructure *pt_map;
GST_INFO ("creating AUX receiver");
bin = gst_bin_new (NULL);
rtx = gst_element_factory_make ("rtprtxreceive", NULL);
pt_map = gst_structure_new ("application/x-rtp-pt-map",
"8", G_TYPE_UINT, 98, "96", G_TYPE_UINT, 99, NULL);
g_object_set (rtx, "payload-type-map", pt_map, NULL);
gst_structure_free (pt_map);
gst_bin_add (GST_BIN (bin), rtx);
pad = gst_element_get_static_pad (rtx, "src");
name = g_strdup_printf ("src_%u", sessid);
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (pad);
pad = gst_element_get_static_pad (rtx, "sink");
name = g_strdup_printf ("sink_%u", sessid);
gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (pad);
return bin;
}
static void
join_session (GstElement * pipeline, GstElement * rtpBin, SessionData * session)
{
GstElement *rtpSrc;
GstElement *rtcpSrc;
GstElement *rtcpSink;
gchar *padName;
guint basePort;
g_print ("Joining session %p\n", session);
session->rtpbin = g_object_ref (rtpBin);
basePort = 5000 + (session->sessionNum * 6);
rtpSrc = gst_element_factory_make ("udpsrc", NULL);
rtcpSrc = gst_element_factory_make ("udpsrc", NULL);
rtcpSink = gst_element_factory_make ("udpsink", NULL);
g_object_set (rtpSrc, "port", basePort, "caps", session->caps, NULL);
g_object_set (rtcpSink, "port", basePort + 5, "host", "127.0.0.1", "sync",
FALSE, "async", FALSE, NULL);
g_object_set (rtcpSrc, "port", basePort + 1, NULL);
g_print ("Connecting to %i/%i/%i\n", basePort, basePort + 1, basePort + 5);
/* enable RFC4588 retransmission handling by setting rtprtxreceive
* as the "aux" element of rtpbin */
g_signal_connect (rtpBin, "request-aux-receiver",
(GCallback) request_aux_receiver, session);
gst_bin_add_many (GST_BIN (pipeline), rtpSrc, rtcpSrc, rtcpSink, NULL);
g_signal_connect_data (rtpBin, "pad-added", G_CALLBACK (handle_new_stream),
session_ref (session), (GClosureNotify) session_unref, 0);
g_signal_connect_data (rtpBin, "request-pt-map", G_CALLBACK (request_pt_map),
session_ref (session), (GClosureNotify) session_unref, 0);
padName = g_strdup_printf ("recv_rtp_sink_%u", session->sessionNum);
gst_element_link_pads (rtpSrc, "src", rtpBin, padName);
g_free (padName);
padName = g_strdup_printf ("recv_rtcp_sink_%u", session->sessionNum);
gst_element_link_pads (rtcpSrc, "src", rtpBin, padName);
g_free (padName);
padName = g_strdup_printf ("send_rtcp_src_%u", session->sessionNum);
gst_element_link_pads (rtpBin, padName, rtcpSink, "sink");
g_free (padName);
session_unref (session);
}
int
main (int argc, char **argv)
{
GstPipeline *pipe;
SessionData *videoSession;
SessionData *audioSession;
GstElement *rtpBin;
GstBus *bus;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
pipe = GST_PIPELINE (gst_pipeline_new (NULL));
bus = gst_element_get_bus (GST_ELEMENT (pipe));
g_signal_connect (bus, "message::error", G_CALLBACK (cb_error), pipe);
g_signal_connect (bus, "message::warning", G_CALLBACK (cb_warning), pipe);
g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
g_signal_connect (bus, "message::eos", G_CALLBACK (cb_eos), NULL);
gst_bus_add_signal_watch (bus);
gst_object_unref (bus);
rtpBin = gst_element_factory_make ("rtpbin", NULL);
gst_bin_add (GST_BIN (pipe), rtpBin);
g_object_set (rtpBin, "latency", 200, "do-retransmission", TRUE,
"rtp-profile", GST_RTP_PROFILE_AVPF, NULL);
videoSession = make_video_session (0);
audioSession = make_audio_session (1);
join_session (GST_ELEMENT (pipe), rtpBin, videoSession);
join_session (GST_ELEMENT (pipe), rtpBin, audioSession);
g_print ("starting client pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_PLAYING);
g_main_loop_run (loop);
g_print ("stoping client pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_NULL);
gst_object_unref (pipe);
g_main_loop_unref (loop);
return 0;
}