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examples: rtp: Add end-to-end rtpbin example with RTX elements

Browse source 
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.
This commit is contained in:
Torrie Fischer 2013-11-13 15:11:35 -05:00 committed by Wim Taymans
parent d2edee4b49
commit e29b5f8b41
4 changed files with 672 additions and 1 deletions
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2
tests/examples/rtp/.gitignore vendored
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@ -1,2 +1,4 @@
client-PCMA
server-alsasrc-PCMA
client-rtpaux
server-rtpaux

11
tests/examples/rtp/Makefile.am
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@ -1,8 +1,17 @@
noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA
noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA \
client-rtpaux server-rtpaux
# FIXME 0.11: ignore GValueArray warnings for now until this is sorted
ERROR_CFLAGS=
server_rtpaux_SOURCES = server-rtpaux.c
server_rtpaux_CFLAGS = $(GST_CFLAGS)
server_rtpaux_LDADD = $(GST_LIBS)
client_rtpaux_SOURCES = client-rtpaux.c
client_rtpaux_CFLAGS = $(GST_CFLAGS)
client_rtpaux_LDADD = $(GST_LIBS)
server_alsasrc_PCMA_SOURCES = server-alsasrc-PCMA.c
server_alsasrc_PCMA_CFLAGS = $(GST_CFLAGS)
server_alsasrc_PCMA_LDADD = $(GST_LIBS) $(LIBM)

369
tests/examples/rtp/client-rtpaux.c Normal file
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/* 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;
}

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/* 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>
/*
* An 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.
*
* In both sessions, we set "rtprtxsend" as the session's "aux" element
* in rtpbin, which enables RFC4588 retransmission for that session.
*
* .-------. .-------. .-------. .------------. .-------.
* |audiots| |alawenc| |pcmapay| | rtpbin | |udpsink|
* | src->sink src->sink src->send_rtp_0 send_rtp_0->sink |
* '-------' '-------' '-------' | | '-------'
* | |
* .-------. .---------. .---------. | | .-------.
* |audiots| |theoraenc| |theorapay| | | |udpsink|
* | src->sink src->sink src->send_rtp_1 send_rtp_1->sink |
* '-------' '---------' '---------' | | '-------'
* | |
* .------. | |
* |udpsrc| | | .-------.
* | src->recv_rtcp_0 | |udpsink|
* '------' | send_rtcp_0->sink |
* | | '-------'
* .------. | |
* |udpsrc| | | .-------.
* | src->recv_rtcp_1 | |udpsink|
* '------' | send_rtcp_1->sink |
* '------------' '-------'
*
* To keep the set of ports consistent across both this server and the
* corresponding client, a SessionData struct maps a rtpbin session number to
* a GstBin and is used to create the corresponding udp sinks with correct
* ports.
*/
typedef struct _SessionData
{
int ref;
guint sessionNum;
GstElement *input;
} 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_free (session);
}
}
static SessionData *
session_new (guint sessionNum)
{
SessionData *ret = g_new0 (SessionData, 1);
ret->sessionNum = sessionNum;
return session_ref (ret);
}
/*
* 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));
}
}
/*
* Creates a GstGhostPad named "src" on the given bin, pointed at the "src" pad
* of the given element
*/
static void
setup_ghost (GstElement * src, GstBin * bin)
{
GstPad *srcPad = gst_element_get_static_pad (src, "src");
GstPad *binPad = gst_ghost_pad_new ("src", srcPad);
gst_element_add_pad (GST_ELEMENT (bin), binPad);
}
static SessionData *
make_audio_session (guint sessionNum)
{
SessionData *session;
GstBin *audioBin = GST_BIN (gst_bin_new (NULL));
GstElement *audioSrc = gst_element_factory_make ("audiotestsrc", NULL);
GstElement *encoder = gst_element_factory_make ("alawenc", NULL);
GstElement *payloader = gst_element_factory_make ("rtppcmapay", NULL);
g_object_set (audioSrc, "is-live", TRUE, NULL);
gst_bin_add_many (audioBin, audioSrc, encoder, payloader, NULL);
gst_element_link_many (audioSrc, encoder, payloader, NULL);
setup_ghost (payloader, audioBin);
session = session_new (sessionNum);
session->input = GST_ELEMENT (audioBin);
return session;
}
static SessionData *
make_video_session (guint sessionNum)
{
GstBin *videoBin = GST_BIN (gst_bin_new (NULL));
GstElement *videoSrc = gst_element_factory_make ("videotestsrc", NULL);
GstElement *encoder = gst_element_factory_make ("theoraenc", NULL);
GstElement *payloader = gst_element_factory_make ("rtptheorapay", NULL);
GstCaps *videoCaps;
SessionData *session;
g_object_set (videoSrc, "is-live", TRUE, "horizontal-speed", 1, NULL);
g_object_set (payloader, "config-interval", 2, NULL);
gst_bin_add_many (videoBin, videoSrc, encoder, payloader, NULL);
videoCaps = gst_caps_new_simple ("video/x-raw",
"width", G_TYPE_INT, 352,
"height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 15, 1, NULL);
gst_element_link_filtered (videoSrc, encoder, videoCaps);
gst_element_link (encoder, payloader);
setup_ghost (payloader, videoBin);
session = session_new (sessionNum);
session->input = GST_ELEMENT (videoBin);
return session;
}
static GstElement *
request_aux_sender (GstElement * rtpbin, guint sessid, SessionData * session)
{
GstElement *rtx, *bin;
GstPad *pad;
gchar *name;
GST_INFO ("creating AUX sender");
bin = gst_bin_new (NULL);
rtx = gst_element_factory_make ("rtprtxsend", NULL);
g_object_set (rtx, "rtx-payload-type", 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;
}
/*
* This function sets up the UDP sinks and sources for RTP/RTCP, adds the
* given session's bin into the pipeline, and links it to the properly numbered
* pads on the rtpbin
*/
static void
add_stream (GstPipeline * pipe, GstElement * rtpBin, SessionData * session)
{
GstElement *rtpSink = gst_element_factory_make ("udpsink", NULL);
GstElement *rtcpSink = gst_element_factory_make ("udpsink", NULL);
GstElement *rtcpSrc = gst_element_factory_make ("udpsrc", NULL);
GstElement *identity = gst_element_factory_make ("identity", NULL);
int basePort;
gchar *padName;
basePort = 5000 + (session->sessionNum * 6);
gst_bin_add_many (GST_BIN (pipe), rtpSink, rtcpSink, rtcpSrc, identity,
session->input, NULL);
/* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
g_signal_connect (rtpBin, "request-aux-sender",
(GCallback) request_aux_sender, session);
g_object_set (rtpSink, "port", basePort, "host", "127.0.0.1", NULL);
g_object_set (rtcpSink, "port", basePort + 1, "host", "127.0.0.1", "sync",
FALSE, "async", FALSE, NULL);
g_object_set (rtcpSrc, "port", basePort + 5, NULL);
/* this is just to drop some rtp packets at random, to demonstrate
* that rtprtxsend actually works */
g_object_set (identity, "drop-probability", 0.01, NULL);
padName = g_strdup_printf ("send_rtp_sink_%u", session->sessionNum);
gst_element_link_pads (session->input, "src", rtpBin, padName);
g_free (padName);
/* link rtpbin to udpsink directly here if you don't want
* artificial packet loss */
padName = g_strdup_printf ("send_rtp_src_%u", session->sessionNum);
gst_element_link_pads (rtpBin, padName, identity, "sink");
gst_element_link (identity, rtpSink);
g_free (padName);
padName = g_strdup_printf ("send_rtcp_src_%u", session->sessionNum);
gst_element_link_pads (rtpBin, padName, rtcpSink, "sink");
g_free (padName);
padName = g_strdup_printf ("recv_rtcp_sink_%u", session->sessionNum);
gst_element_link_pads (rtcpSrc, "src", rtpBin, padName);
g_free (padName);
g_print ("New RTP stream on %i/%i/%i\n", basePort, basePort + 1,
basePort + 5);
session_unref (session);
}
int
main (int argc, char **argv)
{
GstPipeline *pipe;
GstBus *bus;
SessionData *videoSession;
SessionData *audioSession;
GstElement *rtpBin;
GMainLoop *loop;
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::state-changed", G_CALLBACK (cb_state), pipe);
gst_bus_add_signal_watch (bus);
gst_object_unref (bus);
rtpBin = gst_element_factory_make ("rtpbin", NULL);
gst_bin_add (GST_BIN (pipe), rtpBin);
videoSession = make_video_session (0);
audioSession = make_audio_session (1);
add_stream (pipe, rtpBin, videoSession);
add_stream (pipe, rtpBin, audioSession);
g_print ("starting server pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_PLAYING);
g_main_loop_run (loop);
g_print ("stopping server pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_NULL);
gst_object_unref (pipe);
g_main_loop_unref (loop);
return 0;
}