gstreamer/tests/examples/rtp/client-rtpaux.c
Torrie Fischer e29b5f8b41 examples: rtp: Add end-to-end rtpbin example with RTX elements
This example demonstrates how to use rtpbin with retransmission (rtx)
elements set in the place of rtpbin's "aux" elements in order to
enable RTP retransmission according to the rules of RFC4588.
2014-01-03 20:48:29 +01:00

370 lines
12 KiB
C

/* 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 <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;
g_print ("Looking for caps for pt %u in session %u, have %u\n", pt, session,
data->sessionNum);
if (session == data->sessionNum) {
g_print ("Returning %s\n", gst_caps_to_string (data->caps));
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;
gst_bin_add (GST_BIN (gst_element_get_parent (session->rtpbin)),
session->output);
gst_element_sync_state_with_parent (session->output);
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;
GST_INFO ("creating AUX receiver");
bin = gst_bin_new (NULL);
rtx = gst_element_factory_make ("rtprtxreceive", NULL);
g_object_set (rtx, "rtx-payload-types", "99", NULL);
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_emit_by_name (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, 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;
}