mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-02 12:32:29 +00:00
rtpmanagerbad: add RTP streaming elements
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:
parent
da085a3713
commit
3ef737605a
17 changed files with 1697 additions and 1 deletions
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@ -475,6 +475,7 @@ AG_GST_CHECK_PLUGIN(proxy)
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AG_GST_CHECK_PLUGIN(rawparse)
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AG_GST_CHECK_PLUGIN(removesilence)
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AG_GST_CHECK_PLUGIN(rist)
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AG_GST_CHECK_PLUGIN(rtp)
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AG_GST_CHECK_PLUGIN(sdp)
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AG_GST_CHECK_PLUGIN(segmentclip)
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AG_GST_CHECK_PLUGIN(siren)
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@ -2554,6 +2555,7 @@ gst/proxy/Makefile
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gst/rawparse/Makefile
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gst/removesilence/Makefile
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gst/rist/Makefile
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gst/rtp/Makefile
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gst/sdp/Makefile
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gst/segmentclip/Makefile
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gst/siren/Makefile
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@ -8,7 +8,7 @@ foreach plugin : ['accurip', 'adpcmdec', 'adpcmenc', 'aiff', 'asfmux',
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'ivfparse', 'ivtc', 'jp2kdecimator', 'jpegformat', 'librfb',
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'midi', 'mpegdemux', 'mpegpsmux', 'mpegtsdemux', 'mpegtsmux',
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'mxf', 'netsim', 'onvif', 'pcapparse', 'pnm', 'proxy',
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'rawparse', 'removesilence', 'rist', 'sdp', 'segmentclip',
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'rawparse', 'removesilence', 'rist', 'rtp', 'sdp', 'segmentclip',
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'siren', 'smooth', 'speed', 'subenc', 'timecode',
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'videofilters', 'videoframe_audiolevel', 'videoparsers',
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'videosignal', 'vmnc', 'y4m', 'yadif']
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17
gst/rtp/Makefile.am
Normal file
17
gst/rtp/Makefile.am
Normal file
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@ -0,0 +1,17 @@
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plugin_LTLIBRARIES = libgstrtpmanagerbad.la
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libgstrtpmanagerbad_la_SOURCES = \
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gstrtp-utils.c \
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gstrtpsink.c \
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gstrtpsrc.c \
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plugin.c
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libgstrtpmanagerbad_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS)
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libgstrtpmanagerbad_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
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libgstrtpmanagerbad_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = \
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gstrtp-utils.h \
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gstrtpcaps.h \
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gstrtpsink.h \
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gstrtpsrc.h
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39
gst/rtp/gstrtp-utils.c
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39
gst/rtp/gstrtp-utils.c
Normal file
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@ -0,0 +1,39 @@
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/*
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* See: https://bugzilla.gnome.org/show_bug.cgi?id=779765
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*/
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#include "gstrtp-utils.h"
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static void
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gst_rtp_utils_uri_query_foreach (const gchar * key, const gchar * value,
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GObject * src)
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{
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if (key == NULL) {
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GST_WARNING_OBJECT (src, "Refusing to use empty key.");
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return;
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}
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if (value == NULL) {
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GST_WARNING_OBJECT (src, "Refusing to use NULL for key %s.", key);
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return;
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}
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GST_DEBUG_OBJECT (src, "Setting property '%s' to '%s'", key, value);
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gst_util_set_object_arg (src, key, value);
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}
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void
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gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri)
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{
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GHashTable *hash_table;
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g_return_if_fail (uri != NULL);
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hash_table = gst_uri_get_query_table (uri);
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if (hash_table) {
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g_hash_table_foreach (hash_table,
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(GHFunc) gst_rtp_utils_uri_query_foreach, obj);
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g_hash_table_unref (hash_table);
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}
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}
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8
gst/rtp/gstrtp-utils.h
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8
gst/rtp/gstrtp-utils.h
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@ -0,0 +1,8 @@
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#ifndef __GST_RTP_UTILS_H__
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#define __GST_RTP_UTILS_H__
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#include <gst/gst.h>
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void gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri);
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#endif
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581
gst/rtp/gstrtpsink.c
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581
gst/rtp/gstrtpsink.c
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@ -0,0 +1,581 @@
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/* GStreamer
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* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION: gstrtsinkp
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* @title: GstRtpSink
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* @short description: element with Uri interface to stream RTP data to
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* the network.
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*
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* RTP (RFC 3550) is a protocol to stream media over the network while
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* retaining the timing information and providing enough information to
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* reconstruct the correct timing domain by the receiver.
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*
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* The RTP data port should be even, while the RTCP port should be
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* odd. The URI that is entered defines the data port, the RTCP port will
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* be allocated to the next port.
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*
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* This element hooks up the correct sockets to support both RTP as the
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* accompanying RTCP layer.
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*
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* This Bin handles streaming RTP payloaded data on the network.
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*
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* This element also implements the URI scheme `rtp://` allowing to send
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* data on the network by bins that allow use the URI to determine the sink.
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* The RTP URI handler also allows setting properties through the URI query.
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <gio/gio.h>
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#include "gstrtpsink.h"
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#include "gstrtp-utils.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
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#define GST_CAT_DEFAULT gst_rtp_sink_debug
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#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
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#define DEFAULT_PROP_TTL 64
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#define DEFAULT_PROP_TTL_MC 1
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enum
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{
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PROP_0,
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PROP_URI,
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PROP_TTL,
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PROP_TTL_MC,
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PROP_LAST
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};
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static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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#define gst_rtp_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
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#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
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#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
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#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStateChangeReturn
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gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
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static void
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gst_rtp_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpSink *self = GST_RTP_SINK (object);
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switch (prop_id) {
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case PROP_URI:{
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GstUri *uri = NULL;
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GST_RTP_SINK_LOCK (object);
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uri = gst_uri_from_string (g_value_get_string (value));
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if (uri == NULL)
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break;
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if (self->uri)
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gst_uri_unref (self->uri);
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self->uri = uri;
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/* RTP data ports should be even according to RFC 3550, while the
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* RTCP is sent on odd ports. Just warn if there is a mismatch. */
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if (gst_uri_get_port (self->uri) % 2)
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GST_WARNING_OBJECT (self,
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"Port %u is not even, this is not standard (see RFC 3550).",
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gst_uri_get_port (self->uri));
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gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
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GST_RTP_SINK_UNLOCK (object);
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break;
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}
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case PROP_TTL:
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self->ttl = g_value_get_int (value);
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break;
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case PROP_TTL_MC:
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self->ttl_mc = g_value_get_int (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpSink *self = GST_RTP_SINK (object);
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switch (prop_id) {
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case PROP_URI:
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GST_RTP_SINK_LOCK (object);
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if (self->uri)
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g_value_take_string (value, gst_uri_to_string (self->uri));
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else
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g_value_set_string (value, NULL);
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GST_RTP_SINK_UNLOCK (object);
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break;
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case PROP_TTL:
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g_value_set_int (value, self->ttl);
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break;
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case PROP_TTL_MC:
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g_value_set_int (value, self->ttl_mc);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_sink_finalize (GObject * gobject)
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{
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GstRtpSink *self = GST_RTP_SINK (gobject);
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if (self->uri)
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gst_uri_unref (self->uri);
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g_mutex_clear (&self->lock);
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G_OBJECT_CLASS (parent_class)->finalize (gobject);
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}
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static gboolean
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gst_rtp_sink_setup_elements (GstRtpSink * self)
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{
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/*GstPad *pad; */
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GSocket *socket;
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GInetAddress *addr;
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gchar name[48];
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GstCaps *caps;
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/* Should not be NULL */
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g_return_val_if_fail (self->uri != NULL, FALSE);
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/* if not already configured */
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if (self->funnel_rtp == NULL) {
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self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
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if (self->funnel_rtp == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "funnel_rtp element is not available"));
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return FALSE;
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}
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self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
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if (self->funnel_rtcp == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "funnel_rtcp element is not available"));
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return FALSE;
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}
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self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
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if (self->rtp_sink == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "rtp_sink element is not available"));
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return FALSE;
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}
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self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
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if (self->rtcp_src == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "rtcp_src element is not available"));
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return FALSE;
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}
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self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
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if (self->rtcp_sink == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "rtcp_sink element is not available"));
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return FALSE;
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}
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gst_bin_add (GST_BIN (self), self->funnel_rtp);
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gst_bin_add (GST_BIN (self), self->funnel_rtcp);
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/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
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* not all at the same moment */
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g_object_set (self->rtp_sink,
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"host", gst_uri_get_host (self->uri),
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"port", gst_uri_get_port (self->uri),
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"ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
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gst_bin_add (GST_BIN (self), self->rtp_sink);
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g_object_set (self->rtcp_sink,
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"host", gst_uri_get_host (self->uri),
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"port", gst_uri_get_port (self->uri) + 1,
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"ttl", self->ttl, "ttl-mc", self->ttl_mc,
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/* Set false since we're reusing a socket */
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"auto-multicast", FALSE, NULL);
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gst_bin_add (GST_BIN (self), self->rtcp_sink);
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/* no need to set address if unicast */
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caps = gst_caps_new_empty_simple ("application/x-rtcp");
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g_object_set (self->rtcp_src,
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"port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
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gst_caps_unref (caps);
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addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
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if (g_inet_address_get_is_multicast (addr)) {
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g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
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NULL);
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}
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g_object_unref (addr);
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gst_bin_add (GST_BIN (self), self->rtcp_src);
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gst_element_link (self->funnel_rtp, self->rtp_sink);
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gst_element_link (self->funnel_rtcp, self->rtcp_sink);
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gst_element_sync_state_with_parent (self->funnel_rtp);
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gst_element_sync_state_with_parent (self->funnel_rtcp);
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gst_element_sync_state_with_parent (self->rtp_sink);
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gst_element_sync_state_with_parent (self->rtcp_src);
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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: */
|
72
gst/rtp/gstrtpsink.h
Normal file
72
gst/rtp/gstrtpsink.h
Normal file
|
@ -0,0 +1,72 @@
|
|||
/* 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__ */
|
731
gst/rtp/gstrtpsrc.c
Normal file
731
gst/rtp/gstrtpsrc.c
Normal file
|
@ -0,0 +1,731 @@
|
|||
/* 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
76
gst/rtp/gstrtpsrc.h
Normal 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
15
gst/rtp/meson.build
Normal 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
28
gst/rtp/plugin.c
Normal 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);
|
|
@ -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')
|
||||
|
|
|
@ -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
2
tests/check/elements/.gitignore
vendored
|
@ -49,6 +49,8 @@ pcapparse
|
|||
pnm
|
||||
rtponvifparse
|
||||
rtponviftimestamp
|
||||
rtpsrc
|
||||
rtpsink
|
||||
shm
|
||||
srtp
|
||||
templatematch
|
||||
|
|
56
tests/check/elements/rtpsink.c
Normal file
56
tests/check/elements/rtpsink.c
Normal 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
58
tests/check/elements/rtpsrc.c
Normal 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);
|
|
@ -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]],
|
||||
|
|
Loading…
Reference in a new issue