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rtpmanagerbad: add RTP streaming elements

Browse source 
This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.

https://bugzilla.gnome.org/show_bug.cgi?id=703111

The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.

The code can be used as follows

```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink

gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```

rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate

GStreamer 1.16 does not yet support the newer GLib templates, so revert.

rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources

for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.

rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches

beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.

rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even

According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.

rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters

Locking is added because the URI allows to access the properties too.

rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction

In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
This commit is contained in:
Marc Leeman 2019-06-03 20:08:23 +00:00 committed by Nicolas Dufresne
parent da085a3713
commit 3ef737605a
17 changed files with 1697 additions and 1 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
configure.ac
View file

@ -475,6 +475,7 @@ AG_GST_CHECK_PLUGIN(proxy)
AG_GST_CHECK_PLUGIN(rawparse)
AG_GST_CHECK_PLUGIN(removesilence)
AG_GST_CHECK_PLUGIN(rist)
AG_GST_CHECK_PLUGIN(rtp)
AG_GST_CHECK_PLUGIN(sdp)
AG_GST_CHECK_PLUGIN(segmentclip)
AG_GST_CHECK_PLUGIN(siren)
@ -2554,6 +2555,7 @@ gst/proxy/Makefile
gst/rawparse/Makefile
gst/removesilence/Makefile
gst/rist/Makefile
gst/rtp/Makefile
gst/sdp/Makefile
gst/segmentclip/Makefile
gst/siren/Makefile

2
gst/meson.build
View file

@ -8,7 +8,7 @@ foreach plugin : ['accurip', 'adpcmdec', 'adpcmenc', 'aiff', 'asfmux',
'ivfparse', 'ivtc', 'jp2kdecimator', 'jpegformat', 'librfb',
'midi', 'mpegdemux', 'mpegpsmux', 'mpegtsdemux', 'mpegtsmux',
'mxf', 'netsim', 'onvif', 'pcapparse', 'pnm', 'proxy',
'rawparse', 'removesilence', 'rist', 'sdp', 'segmentclip',
'rawparse', 'removesilence', 'rist', 'rtp', 'sdp', 'segmentclip',
'siren', 'smooth', 'speed', 'subenc', 'timecode',
'videofilters', 'videoframe_audiolevel', 'videoparsers',
'videosignal', 'vmnc', 'y4m', 'yadif']

17
gst/rtp/Makefile.am Normal file
View file

@ -0,0 +1,17 @@
plugin_LTLIBRARIES = libgstrtpmanagerbad.la
libgstrtpmanagerbad_la_SOURCES = \
gstrtp-utils.c \
gstrtpsink.c \
gstrtpsrc.c \
plugin.c
libgstrtpmanagerbad_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstrtpmanagerbad_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
libgstrtpmanagerbad_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = \
gstrtp-utils.h \
gstrtpcaps.h \
gstrtpsink.h \
gstrtpsrc.h

39
gst/rtp/gstrtp-utils.c Normal file
View file

@ -0,0 +1,39 @@
/*
* See: https://bugzilla.gnome.org/show_bug.cgi?id=779765
*/
#include "gstrtp-utils.h"
static void
gst_rtp_utils_uri_query_foreach (const gchar * key, const gchar * value,
GObject * src)
{
if (key == NULL) {
GST_WARNING_OBJECT (src, "Refusing to use empty key.");
return;
}
if (value == NULL) {
GST_WARNING_OBJECT (src, "Refusing to use NULL for key %s.", key);
return;
}
GST_DEBUG_OBJECT (src, "Setting property '%s' to '%s'", key, value);
gst_util_set_object_arg (src, key, value);
}
void
gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri)
{
GHashTable *hash_table;
g_return_if_fail (uri != NULL);
hash_table = gst_uri_get_query_table (uri);
if (hash_table) {
g_hash_table_foreach (hash_table,
(GHFunc) gst_rtp_utils_uri_query_foreach, obj);
g_hash_table_unref (hash_table);
}
}

8
gst/rtp/gstrtp-utils.h Normal file
View file

@ -0,0 +1,8 @@
#ifndef __GST_RTP_UTILS_H__
#define __GST_RTP_UTILS_H__
#include <gst/gst.h>
void gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri);
#endif

581
gst/rtp/gstrtpsink.c Normal file
View file

@ -0,0 +1,581 @@
/* GStreamer
* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.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.
*/
/**
* SECTION: gstrtsinkp
* @title: GstRtpSink
* @short description: element with Uri interface to stream RTP data to
* the network.
*
* RTP (RFC 3550) is a protocol to stream media over the network while
* retaining the timing information and providing enough information to
* reconstruct the correct timing domain by the receiver.
*
* The RTP data port should be even, while the RTCP port should be
* odd. The URI that is entered defines the data port, the RTCP port will
* be allocated to the next port.
*
* This element hooks up the correct sockets to support both RTP as the
* accompanying RTCP layer.
*
* This Bin handles streaming RTP payloaded data on the network.
*
* This element also implements the URI scheme `rtp://` allowing to send
* data on the network by bins that allow use the URI to determine the sink.
* The RTP URI handler also allows setting properties through the URI query.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gio/gio.h>
#include "gstrtpsink.h"
#include "gstrtp-utils.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
#define GST_CAT_DEFAULT gst_rtp_sink_debug
#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
#define DEFAULT_PROP_TTL 64
#define DEFAULT_PROP_TTL_MC 1
enum
{
PROP_0,
PROP_URI,
PROP_TTL,
PROP_TTL_MC,
PROP_LAST
};
static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
gpointer iface_data);
#define gst_rtp_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStateChangeReturn
gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
static void
gst_rtp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpSink *self = GST_RTP_SINK (object);
switch (prop_id) {
case PROP_URI:{
GstUri *uri = NULL;
GST_RTP_SINK_LOCK (object);
uri = gst_uri_from_string (g_value_get_string (value));
if (uri == NULL)
break;
if (self->uri)
gst_uri_unref (self->uri);
self->uri = uri;
/* RTP data ports should be even according to RFC 3550, while the
* RTCP is sent on odd ports. Just warn if there is a mismatch. */
if (gst_uri_get_port (self->uri) % 2)
GST_WARNING_OBJECT (self,
"Port %u is not even, this is not standard (see RFC 3550).",
gst_uri_get_port (self->uri));
gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
GST_RTP_SINK_UNLOCK (object);
break;
}
case PROP_TTL:
self->ttl = g_value_get_int (value);
break;
case PROP_TTL_MC:
self->ttl_mc = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpSink *self = GST_RTP_SINK (object);
switch (prop_id) {
case PROP_URI:
GST_RTP_SINK_LOCK (object);
if (self->uri)
g_value_take_string (value, gst_uri_to_string (self->uri));
else
g_value_set_string (value, NULL);
GST_RTP_SINK_UNLOCK (object);
break;
case PROP_TTL:
g_value_set_int (value, self->ttl);
break;
case PROP_TTL_MC:
g_value_set_int (value, self->ttl_mc);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_sink_finalize (GObject * gobject)
{
GstRtpSink *self = GST_RTP_SINK (gobject);
if (self->uri)
gst_uri_unref (self->uri);
g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (gobject);
}
static gboolean
gst_rtp_sink_setup_elements (GstRtpSink * self)
{
/*GstPad *pad; */
GSocket *socket;
GInetAddress *addr;
gchar name[48];
GstCaps *caps;
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
/* if not already configured */
if (self->funnel_rtp == NULL) {
self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtp == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "funnel_rtp element is not available"));
return FALSE;
}
self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtcp == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "funnel_rtcp element is not available"));
return FALSE;
}
self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtp_sink element is not available"));
return FALSE;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_src element is not available"));
return FALSE;
}
self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtcp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_sink element is not available"));
return FALSE;
}
gst_bin_add (GST_BIN (self), self->funnel_rtp);
gst_bin_add (GST_BIN (self), self->funnel_rtcp);
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
* not all at the same moment */
g_object_set (self->rtp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri),
"ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
gst_bin_add (GST_BIN (self), self->rtp_sink);
g_object_set (self->rtcp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri) + 1,
"ttl", self->ttl, "ttl-mc", self->ttl_mc,
/* Set false since we're reusing a socket */
"auto-multicast", FALSE, NULL);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src,
"port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
gst_caps_unref (caps);
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (g_inet_address_get_is_multicast (addr)) {
g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
NULL);
}
g_object_unref (addr);
gst_bin_add (GST_BIN (self), self->rtcp_src);
gst_element_link (self->funnel_rtp, self->rtp_sink);
gst_element_link (self->funnel_rtcp, self->rtcp_sink);
gst_element_sync_state_with_parent (self->funnel_rtp);
gst_element_sync_state_with_parent (self->funnel_rtcp);
gst_element_sync_state_with_parent (self->rtp_sink);
gst_element_sync_state_with_parent (self->rtcp_src);
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
gst_element_sync_state_with_parent (self->rtcp_sink);
}
/* pads are all named */
g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
return TRUE;
}
static GstPad *
gst_rtp_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstPad *pad = NULL;
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return NULL;
}
if (gst_rtp_sink_setup_elements (self) == FALSE)
return NULL;
GST_RTP_SINK_LOCK (self);
pad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_%u");
g_return_val_if_fail (pad != NULL, NULL);
GST_RTP_SINK_UNLOCK (self);
return pad;
}
static void
gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
GST_RTP_SINK_LOCK (self);
gst_element_release_request_pad (self->rtpbin, rpad);
gst_object_unref (rpad);
gst_pad_set_active (pad, FALSE);
gst_element_remove_pad (GST_ELEMENT (self), pad);
GST_RTP_SINK_UNLOCK (self);
}
static void
gst_rtp_sink_class_init (GstRtpSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->set_property = gst_rtp_sink_set_property;
gobject_class->get_property = gst_rtp_sink_get_property;
gobject_class->finalize = gst_rtp_sink_finalize;
gstelement_class->change_state = gst_rtp_sink_change_state;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
/**
* GstRtpSink:uri:
*
* uri to stream RTP to. All GStreamer parameters can be
* encoded in the URI, this URI format is RFC compliant.
*/
g_object_class_install_property (gobject_class, PROP_URI,
g_param_spec_string ("uri", "URI",
"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:ttl:
*
* Set the unicast TTL parameter.
*/
g_object_class_install_property (gobject_class, PROP_TTL,
g_param_spec_int ("ttl", "Unicast TTL",
"Used for setting the unicast TTL parameter",
0, 255, DEFAULT_PROP_TTL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:ttl-mc:
*
* Set the multicast TTL parameter.
*/
g_object_class_install_property (gobject_class, PROP_TTL_MC,
g_param_spec_int ("ttl-mc", "Multicast TTL",
"Used for setting the multicast TTL parameter", 0, 255,
DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP Sink element",
"Generic/Bin/Sink",
"Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
}
static void
gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
GstElement * new_element, gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
new_element);
}
static void
gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GstCaps *caps = gst_pad_query_caps (pad, NULL);
GstPad *upad;
/* Expose RTP data pad only */
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
GST_PTR_FORMAT ".", element, pad, caps);
/* Sanity checks */
if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
/* Src pad, do not expose */
gst_caps_unref (caps);
return;
}
if (G_LIKELY (caps)) {
GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
if (gst_caps_can_intersect (caps, ref_caps)) {
/* SRC RTCP caps, do not expose */
gst_caps_unref (ref_caps);
gst_caps_unref (caps);
return;
}
gst_caps_unref (ref_caps);
} else {
GST_ERROR_OBJECT (self, "Pad with no caps detected.");
gst_caps_unref (caps);
return;
}
gst_caps_unref (caps);
upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
if (upad == NULL) {
GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
gst_caps_unref (caps);
return;
}
GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
gst_pad_link (pad, upad);
gst_object_unref (upad);
}
static void
gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
pad);
}
static gboolean
gst_rtp_sink_setup_rtpbin (GstRtpSink * self)
{
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return FALSE;
}
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "element-added",
G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
gst_bin_add (GST_BIN (self), self->rtpbin);
gst_element_sync_state_with_parent (self->rtpbin);
return TRUE;
}
static GstStateChangeReturn
gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GST_DEBUG_OBJECT (self, "changing state: %s => %s",
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
default:
break;
}
return ret;
}
static void
gst_rtp_sink_init (GstRtpSink * self)
{
self->rtpbin = NULL;
self->funnel_rtp = NULL;
self->funnel_rtcp = NULL;
self->rtp_sink = NULL;
self->rtcp_src = NULL;
self->rtcp_sink = NULL;
self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
self->ttl = DEFAULT_PROP_TTL;
self->ttl_mc = DEFAULT_PROP_TTL_MC;
if (gst_rtp_sink_setup_rtpbin (self) == FALSE)
return;
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
gst_bin_set_suppressed_flags (GST_BIN (self),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
g_mutex_init (&self->lock);
}
static guint
gst_rtp_sink_uri_get_type (GType type)
{
return GST_URI_SINK;
}
static const gchar *const *
gst_rtp_sink_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { (char *) "rtp", NULL };
return protocols;
}
static gchar *
gst_rtp_sink_uri_get_uri (GstURIHandler * handler)
{
GstRtpSink *self = (GstRtpSink *) handler;
return gst_uri_to_string (self->uri);
}
static gboolean
gst_rtp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRtpSink *self = (GstRtpSink *) handler;
g_object_set (G_OBJECT (self), "uri", uri, NULL);
return TRUE;
}
static void
gst_rtp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtp_sink_uri_get_type;
iface->get_protocols = gst_rtp_sink_uri_get_protocols;
iface->get_uri = gst_rtp_sink_uri_get_uri;
iface->set_uri = gst_rtp_sink_uri_set_uri;
}
/* ex: set tabstop=2 shiftwidth=2 expandtab: */

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/* GStreamer
* Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.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.
*/
#ifndef __GST_RTP_SINK_H__
#define __GST_RTP_SINK_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_SINK \
(gst_rtp_sink_get_type())
#define GST_RTP_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SINK, GstRtpSink))
#define GST_RTP_SINK_CAST(obj) \
((GstRtpSink *) obj)
#define GST_RTP_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SINK, GstRtpSinkClass))
#define GST_IS_RTP_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SINK))
#define GST_IS_RTP_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SINK))
typedef struct _GstRtpSink GstRtpSink;
typedef struct _GstRtpSinkClass GstRtpSinkClass;
struct _GstRtpSink
{
GstBin parent;
GstBin parent_instance;
/* Properties */
GstUri *uri;
gint ttl;
gint ttl_mc;
/* Internal elements */
GstElement *rtpbin;
GstElement *funnel_rtp;
GstElement *funnel_rtcp;
GstElement *rtp_sink;
GstElement *rtcp_src;
GstElement *rtcp_sink;
GMutex lock;
};
struct _GstRtpSinkClass
{
GstBinClass parent;
};
GType gst_rtp_sink_get_type (void);
G_END_DECLS
#endif /* __GST_RTP_SINK_H__ */

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/* GStreamer
* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.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.
*/
/**
* SECTION: gstrtpsrc
* @title: GstRtpSrc
* @short description: element with Uri interface to get RTP data from
* the network.
*
* RTP (RFC 3550) is a protocol to stream media over the network while
* retaining the timing information and providing enough information to
* reconstruct the correct timing domain by the receiver.
*
* The RTP data port should be even, while the RTCP port should be
* odd. The URI that is entered defines the data port, the RTCP port will
* be allocated to the next port.
*
* This element hooks up the correct sockets to support both RTP as the
* accompanying RTCP layer.
*
* This Bin handles taking in of data from the network and provides the
* RTP payloaded data.
*
* This element also implements the URI scheme `rtp://` allowing to render
* RTP streams in GStreamer based media players. The RTP URI handler also
* allows setting properties through the URI query.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/net/net.h>
#include <gst/rtp/gstrtppayloads.h>
#include "gstrtpsrc.h"
#include "gstrtp-utils.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
#define GST_CAT_DEFAULT gst_rtp_src_debug
#define DEFAULT_PROP_TTL 64
#define DEFAULT_PROP_TTL_MC 1
#define DEFAULT_PROP_ENCODING_NAME NULL
#define DEFAULT_PROP_LATENCY 200
#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
enum
{
PROP_0,
PROP_URI,
PROP_TTL,
PROP_TTL_MC,
PROP_ENCODING_NAME,
PROP_LATENCY,
PROP_LAST
};
static void gst_rtp_src_uri_handler_init (gpointer g_iface,
gpointer iface_data);
#define gst_rtp_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init);
GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source"));
#define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock)
#define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj)))
#define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj)))
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStateChangeReturn
gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
/**
* gst_rtp_src_rtpbin_erquest_pt_map_cb:
* @self: The current #GstRtpSrc object
*
* #GstRtpBin callback to map a pt on RTP caps.
*
* Returns: (transfer none): the guess on the RTP caps based on the PT
* and caps.
*/
static GstCaps *
gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id,
guint pt, gpointer data)
{
GstRtpSrc *self = GST_RTP_SRC (data);
const GstRTPPayloadInfo *p = NULL;
GST_DEBUG_OBJECT (self,
"Requesting caps for session-id 0x%x and pt %u.", session_id, pt);
/* the encoding-name has more relevant information */
if (self->encoding_name != NULL) {
/* Unfortunately, the media needs to be passed in the function. Since
* it is not known, try for video if video not found. */
p = gst_rtp_payload_info_for_name ("video", self->encoding_name);
if (p == NULL)
p = gst_rtp_payload_info_for_name ("audio", self->encoding_name);
}
/* Static payload types, this is a simple lookup */
if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
p = gst_rtp_payload_info_for_pt (pt);
}
if (p != NULL) {
GstCaps *ret = gst_caps_new_simple ("application/x-rtp",
"encoding-name", G_TYPE_STRING, p->encoding_name,
"clock-rate", G_TYPE_INT, p->clock_rate,
"media", G_TYPE_STRING, p->media, NULL);
GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret);
return ret;
}
GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and"
" the encoding-name was not set.");
return NULL;
}
static void
gst_rtp_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpSrc *self = GST_RTP_SRC (object);
GstCaps *caps;
switch (prop_id) {
case PROP_URI:{
GstUri *uri = NULL;
GST_RTP_SRC_LOCK (object);
uri = gst_uri_from_string (g_value_get_string (value));
if (uri == NULL)
break;
if (self->uri)
gst_uri_unref (self->uri);
self->uri = uri;
if (gst_uri_get_port (self->uri) % 2)
GST_WARNING_OBJECT (self,
"Port %u is not even, this is not standard (see RFC 3550).",
gst_uri_get_port (self->uri));
gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
GST_RTP_SRC_UNLOCK (object);
break;
}
case PROP_TTL:
self->ttl = g_value_get_int (value);
break;
case PROP_TTL_MC:
self->ttl_mc = g_value_get_int (value);
break;
case PROP_ENCODING_NAME:
g_free (self->encoding_name);
self->encoding_name = g_value_dup_string (value);
if (self->rtp_src) {
caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self);
g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL);
gst_caps_unref (caps);
}
break;
case PROP_LATENCY:
self->latency = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpSrc *self = GST_RTP_SRC (object);
switch (prop_id) {
case PROP_URI:
GST_RTP_SRC_LOCK (object);
if (self->uri)
g_value_take_string (value, gst_uri_to_string (self->uri));
else
g_value_set_string (value, NULL);
GST_RTP_SRC_UNLOCK (object);
break;
case PROP_TTL:
g_value_set_int (value, self->ttl);
break;
case PROP_TTL_MC:
g_value_set_int (value, self->ttl_mc);
break;
case PROP_ENCODING_NAME:
g_value_set_string (value, self->encoding_name);
break;
case PROP_LATENCY:
g_value_set_uint (value, self->latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_src_finalize (GObject * gobject)
{
GstRtpSrc *self = GST_RTP_SRC (gobject);
if (self->uri)
gst_uri_unref (self->uri);
g_free (self->encoding_name);
g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (gobject);
}
static void
gst_rtp_src_class_init (GstRtpSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->set_property = gst_rtp_src_set_property;
gobject_class->get_property = gst_rtp_src_get_property;
gobject_class->finalize = gst_rtp_src_finalize;
gstelement_class->change_state = gst_rtp_src_change_state;
/**
* GstRtpSrc:uri:
*
* uri to an RTP from. All GStreamer parameters can be
* encoded in the URI, this URI format is RFC compliant.
*/
g_object_class_install_property (gobject_class, PROP_URI,
g_param_spec_string ("uri", "URI",
"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:ttl:
*
* Set the unicast TTL parameter. In RTP this of importance for RTCP.
*/
g_object_class_install_property (gobject_class, PROP_TTL,
g_param_spec_int ("ttl", "Unicast TTL",
"Used for setting the unicast TTL parameter",
0, 255, DEFAULT_PROP_TTL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:ttl-mc:
*
* Set the multicast TTL parameter. In RTP this of importance for RTCP.
*/
g_object_class_install_property (gobject_class, PROP_TTL_MC,
g_param_spec_int ("ttl-mc", "Multicast TTL",
"Used for setting the multicast TTL parameter", 0, 255,
DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:encoding-name:
*
* Set the encoding name of the stream to use. This is a short-hand for
* the full caps and maps typically to the encoding-name in the RTP caps.
*/
g_object_class_install_property (gobject_class, PROP_ENCODING_NAME,
g_param_spec_string ("encoding-name", "Caps encoding name",
"Encoding name use to determine caps parameters",
DEFAULT_PROP_ENCODING_NAME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:latency:
*
* Set the size of the latency buffer in the
* GstRtpBin/GstRtpJitterBuffer to compensate for network jitter.
*/
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint ("latency", "Buffer latency in ms",
"Default amount of ms to buffer in the jitterbuffers", 0,
G_MAXUINT, DEFAULT_PROP_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP Source element",
"Generic/Bin/Src",
"Simple RTP src", "Marc Leeman <marc.leeman@gmail.com>");
}
static void
gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSrc *self = GST_RTP_SRC (data);
GstCaps *caps = gst_pad_query_caps (pad, NULL);
GstPad *upad;
gchar name[48];
/* Expose RTP data pad only */
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
GST_PTR_FORMAT ".", element, pad, caps);
/* Sanity checks */
if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
/* Sink pad, do not expose */
gst_caps_unref (caps);
return;
}
if (G_LIKELY (caps)) {
GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
if (gst_caps_can_intersect (caps, ref_caps)) {
/* SRC RTCP caps, do not expose */
gst_caps_unref (ref_caps);
gst_caps_unref (caps);
return;
}
gst_caps_unref (ref_caps);
} else {
GST_ERROR_OBJECT (self, "Pad with no caps detected.");
gst_caps_unref (caps);
return;
}
gst_caps_unref (caps);
GST_RTP_SRC_LOCK (self);
g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads);
upad = gst_ghost_pad_new (name, pad);
gst_pad_set_active (upad, TRUE);
gst_element_add_pad (GST_ELEMENT (self), upad);
GST_RTP_SRC_UNLOCK (self);
}
static void
gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSrc *self = GST_RTP_SRC (data);
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
pad);
}
static void
gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id,
guint ssrc, gpointer data)
{
GstRtpSrc *self = GST_RTP_SRC (data);
GST_INFO_OBJECT (self,
"Dectected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id,
ssrc);
}
static void
gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id,
guint ssrc, gpointer data)
{
GstRtpSrc *self = GST_RTP_SRC (data);
GST_INFO_OBJECT (self, "Dectected a new SSRC: session-id 0x%x, ssrc 0x%x.",
session_id, ssrc);
}
static GstPadProbeReturn
gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info,
gpointer user_data)
{
GstRtpSrc *self = GST_RTP_SRC (user_data);
GstBuffer *buffer;
GstNetAddressMeta *meta;
if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
GstBufferList *buffer_list = info->data;
buffer = gst_buffer_list_get (buffer_list, 0);
} else {
buffer = info->data;
}
meta = gst_buffer_get_net_address_meta (buffer);
GST_OBJECT_LOCK (self);
g_clear_object (&self->rtcp_send_addr);
self->rtcp_send_addr = g_object_ref (meta->addr);
GST_OBJECT_UNLOCK (self);
return GST_PAD_PROBE_OK;
}
static inline void
gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer)
{
GST_OBJECT_LOCK (self);
if (self->rtcp_send_addr)
gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr);
GST_OBJECT_UNLOCK (self);
}
static GstPadProbeReturn
gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
gpointer user_data)
{
GstRtpSrc *self = GST_RTP_SRC (user_data);
if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
GstBufferList *buffer_list = info->data;
GstBuffer *buffer;
gint i;
info->data = buffer_list = gst_buffer_list_make_writable (buffer_list);
for (i = 0; i < gst_buffer_list_length (buffer_list); i++) {
buffer = gst_buffer_list_get (buffer_list, i);
gst_rtp_src_attach_net_address_meta (self, buffer);
}
} else {
GstBuffer *buffer = info->data;
info->data = buffer = gst_buffer_make_writable (buffer);
gst_rtp_src_attach_net_address_meta (self, buffer);
}
return GST_PAD_PROBE_OK;
}
static gboolean
gst_rtp_src_setup_elements (GstRtpSrc * self)
{
GstPad *pad;
GSocket *socket;
GInetAddress *addr;
gchar name[48];
GstCaps *caps;
gchar *address;
guint rtcp_port;
/* Construct the RTP receiver pipeline.
*
* udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
* | rtpbin |
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
*
* This pipeline is fixed for now, note that optionally an FEC stream could
* be added later.
*/
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return FALSE;
}
self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtp_src element is not available"));
return FALSE;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_src element is not available"));
return FALSE;
}
self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
if (self->rtcp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_sink element is not available"));
return FALSE;
}
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
g_signal_connect (self->rtpbin, "request-pt-map",
G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
g_signal_connect (self->rtpbin, "on-new-ssrc",
G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
g_signal_connect (self->rtpbin, "on-ssrc-collision",
G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
g_object_set (self->rtpbin, "latency", self->latency, NULL);
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
* not all at the same moment */
gst_bin_add (GST_BIN (self), self->rtpbin);
gst_bin_add (GST_BIN (self), self->rtp_src);
g_object_set (self->rtp_src,
"address", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri), NULL);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src,
"port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
gst_caps_unref (caps);
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (g_inet_address_get_is_multicast (addr)) {
g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
NULL);
}
g_object_unref (addr);
g_object_set (self->rtcp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri) + 1,
"ttl", self->ttl, "ttl-mc", self->ttl_mc,
/* Set false since we're reusing a socket */
"auto-multicast", FALSE, NULL);
gst_bin_add (GST_BIN (self), self->rtcp_src);
/* share the socket created by the source */
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket,
"address", &address, "port", &rtcp_port, NULL);
addr = g_inet_address_new_from_string (address);
g_free (address);
if (g_inet_address_get_is_multicast (addr)) {
/* mc-ttl is not supported by dynudpsink */
g_socket_set_multicast_ttl (socket, self->ttl_mc);
/* In multicast, send RTCP to the multicast group */
self->rtcp_send_addr = g_inet_socket_address_new (addr, rtcp_port);
} else {
/* In unicast, send RTCP to the detected sender address */
pad = gst_element_get_static_pad (self->rtcp_src, "src");
self->rtcp_recv_probe = gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
gst_rtp_src_on_recv_rtcp, self, NULL);
gst_object_unref (pad);
}
g_object_unref (addr);
pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
self->rtcp_send_probe = gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
gst_rtp_src_on_send_rtcp, self, NULL);
gst_object_unref (pad);
g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
/* pads are all named */
g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
gst_element_sync_state_with_parent (self->rtpbin);
gst_element_sync_state_with_parent (self->rtp_src);
gst_element_sync_state_with_parent (self->rtcp_sink);
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
gst_element_sync_state_with_parent (self->rtcp_src);
return TRUE;
}
static void
gst_rtp_src_stop (GstRtpSrc * self)
{
GstPad *pad;
if (self->rtcp_recv_probe) {
pad = gst_element_get_static_pad (self->rtcp_src, "src");
gst_pad_remove_probe (pad, self->rtcp_recv_probe);
self->rtcp_recv_probe = 0;
gst_object_unref (pad);
}
pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
gst_pad_remove_probe (pad, self->rtcp_send_probe);
self->rtcp_send_probe = 0;
gst_object_unref (pad);
}
static GstStateChangeReturn
gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
{
GstRtpSrc *self = GST_RTP_SRC (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GST_DEBUG_OBJECT (self, "Changing state: %s => %s",
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (gst_rtp_src_setup_elements (self) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtp_src_stop (self);
break;
default:
break;
}
return ret;
}
static void
gst_rtp_src_init (GstRtpSrc * self)
{
self->rtpbin = NULL;
self->rtp_src = NULL;
self->rtcp_src = NULL;
self->rtcp_sink = NULL;
self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
self->ttl = DEFAULT_PROP_TTL;
self->ttl_mc = DEFAULT_PROP_TTL_MC;
self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
self->latency = DEFAULT_PROP_LATENCY;
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
gst_bin_set_suppressed_flags (GST_BIN (self),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
g_mutex_init (&self->lock);
}
static guint
gst_rtp_src_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static const gchar *const *
gst_rtp_src_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { (char *) "rtp", NULL };
return protocols;
}
static gchar *
gst_rtp_src_uri_get_uri (GstURIHandler * handler)
{
GstRtpSrc *self = (GstRtpSrc *) handler;
return gst_uri_to_string (self->uri);
}
static gboolean
gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRtpSrc *self = (GstRtpSrc *) handler;
g_object_set (G_OBJECT (self), "uri", uri, NULL);
return TRUE;
}
static void
gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtp_src_uri_get_type;
iface->get_protocols = gst_rtp_src_uri_get_protocols;
iface->get_uri = gst_rtp_src_uri_get_uri;
iface->set_uri = gst_rtp_src_uri_set_uri;
}
/* ex: set tabstop=2 shiftwidth=2 expandtab: */

76
gst/rtp/gstrtpsrc.h Normal file
View file

@ -0,0 +1,76 @@
/* GStreamer
* Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.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.
*/
#ifndef __GST_RTP_SRC_H__
#define __GST_RTP_SRC_H__
#include <gio/gio.h>
#include <gst/gst.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_SRC \
(gst_rtp_src_get_type())
#define GST_RTP_SRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SRC, GstRtpSrc))
#define GST_RTP_SRC_CAST(obj) \
((GstRtpSrc *) obj)
#define GST_RTP_SRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SRC, GstRtpSrcClass))
#define GST_IS_RTP_SRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SRC))
#define GST_IS_RTP_SRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SRC))
typedef struct _GstRtpSrc GstRtpSrc;
typedef struct _GstRtpSrcClass GstRtpSrcClass;
struct _GstRtpSrc
{
GstBin parent;
/* Properties */
GstUri *uri;
gint ttl;
gint ttl_mc;
gint latency;
gchar *encoding_name;
guint latency_ms;
/* Internal elements */
GstElement *rtpbin;
GstElement *rtp_src;
GstElement *rtcp_src;
GstElement *rtcp_sink;
gulong rtcp_recv_probe;
gulong rtcp_send_probe;
GSocketAddress *rtcp_send_addr;
GMutex lock;
};
struct _GstRtpSrcClass
{
GstBinClass parent;
};
GType gst_rtp_src_get_type (void);
G_END_DECLS
#endif /* __GST_RTP_SRC_H__ */

15
gst/rtp/meson.build Normal file
View file

@ -0,0 +1,15 @@
gst_plugins_rtp_sources = [
'plugin.c',
'gstrtpsink.c',
'gstrtpsrc.c',
'gstrtp-utils.c',
]
gstrtp = library('gstrtpmanagerbad',
gst_plugins_rtp_sources,
dependencies: [gio_dep, gst_dep, gstbase_dep, gstrtp_dep, gstnet_dep, gstcontroller_dep],
include_directories: [configinc],
install: true,
c_args: gst_plugins_bad_args,
install_dir: plugins_install_dir,
)

28
gst/rtp/plugin.c Normal file
View file

@ -0,0 +1,28 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpsink.h"
#include "gstrtpsrc.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
gboolean ret = FALSE;
ret |= gst_element_register (plugin, "rtpsrc",
GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SRC);
ret |= gst_element_register (plugin, "rtpsink",
GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SINK);
return ret;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
rtpmanagerbad,
"GStreamer RTP Plugins",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);

1
meson_options.txt
View file

@ -49,6 +49,7 @@ option('proxy', type : 'feature', value : 'auto')
option('rawparse', type : 'feature', value : 'auto')
option('removesilence', type : 'feature', value : 'auto')
option('rist', type : 'feature', value : 'auto')
option('rtp', type : 'feature', value : 'auto')
option('sdp', type : 'feature', value : 'auto')
option('segmentclip', type : 'feature', value : 'auto')
option('siren', type : 'feature', value : 'auto')

8
tests/check/Makefile.am
View file

@ -290,6 +290,8 @@ check_PROGRAMS = \
elements/pnm \
elements/rtponvifparse \
elements/rtponviftimestamp \
elements/rtpsrc \
elements/rtpsink \
elements/id3mux \
pipelines/mxf \
libs/isoff \
@ -578,6 +580,12 @@ elements_rtponvifparse_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_L
elements_rtponviftimestamp_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_rtponviftimestamp_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
elements_rtpsrc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_rtpsrc_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
elements_rtpsink_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_rtpsink_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
EXTRA_DIST = gst-plugins-bad.supp $(uvch264_dist_data)
orc_bayer_CFLAGS = $(ORC_CFLAGS)

2
tests/check/elements/.gitignore vendored
View file

@ -49,6 +49,8 @@ pcapparse
pnm
rtponvifparse
rtponviftimestamp
rtpsrc
rtpsink
shm
srtp
templatematch

56
tests/check/elements/rtpsink.c Normal file
View file

@ -0,0 +1,56 @@
/* GStreamer
* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.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/check/gstcheck.h>
GST_START_TEST (test_uri_to_properties)
{
GstElement *rtpsink;
gint ttl, ttl_mc;
rtpsink = gst_element_factory_make ("rtpsink", NULL);
/* Sets properties to non-default values (make sure this stays in sync) */
g_object_set (rtpsink, "uri", "rtp://1.230.1.2?" "ttl=8" "&ttl-mc=9", NULL);
g_object_get (rtpsink, "ttl", &ttl, "ttl_mc", &ttl_mc, NULL);
/* Make sure these values are in sync with the one from the URI. */
g_assert_cmpint (ttl, ==, 8);
g_assert_cmpint (ttl_mc, ==, 9);
gst_object_unref (rtpsink);
}
GST_END_TEST;
static Suite *
rtpsink_suite (void)
{
Suite *s = suite_create ("rtpsink");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_uri_to_properties);
return s;
}
GST_CHECK_MAIN (rtpsink);

58
tests/check/elements/rtpsrc.c Normal file
View file

@ -0,0 +1,58 @@
/* GStreamer
* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.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/check/gstcheck.h>
GST_START_TEST (test_uri_to_properties)
{
GstElement *rtpsrc;
guint latency, ttl, ttl_mc;
rtpsrc = gst_element_factory_make ("rtpsrc", NULL);
/* Sets properties to non-default values (make sure this stays in sync) */
g_object_set (rtpsrc, "uri", "rtp://1.230.1.2?"
"latency=300" "&ttl=8" "&ttl-mc=9", NULL);
g_object_get (rtpsrc,
"latency", &latency, "ttl-mc", &ttl_mc, "ttl", &ttl, NULL);
/* Make sure these values are in sync with the one from the URI. */
g_assert_cmpuint (latency, ==, 300);
g_assert_cmpint (ttl, ==, 8);
g_assert_cmpint (ttl_mc, ==, 9);
gst_object_unref (rtpsrc);
}
GST_END_TEST;
static Suite *
rtpsrc_suite (void)
{
Suite *s = suite_create ("rtpsrc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_uri_to_properties);
return s;
}
GST_CHECK_MAIN (rtpsrc);

2
tests/check/meson.build
View file

@ -47,6 +47,8 @@ base_tests = [
[['elements/pnm.c']],
[['elements/rtponvifparse.c']],
[['elements/rtponviftimestamp.c']],
[['elements/rtpsrc.c']],
[['elements/rtpsink.c']],
[['elements/videoframe-audiolevel.c']],
[['elements/viewfinderbin.c']],
[['libs/h264parser.c'], false, [gstcodecparsers_dep]],