mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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rtpstreampay: Add RFC4571 RTP stream payloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
This commit is contained in:
Sebastian Dröge
parent
1fd094d96b
commit
c5284dc047
4 changed files with 349 additions and 2 deletions
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@ -78,7 +78,8 @@ libgstrtp_la_SOURCES = \
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gstrtpvp8depay.c \
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gstrtpvp8pay.c \
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gstrtpvrawdepay.c \
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gstrtpvrawpay.c
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gstrtpvrawpay.c \
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gstrtpstreampay.c
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libgstrtp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
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$(GST_CFLAGS) -Dvp8_norm=gst_rtpvp8_vp8_norm \
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@ -173,7 +174,8 @@ noinst_HEADERS = \
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gstrtpvp8depay.h \
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gstrtpvp8pay.h \
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gstrtpvrawdepay.h \
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gstrtpvrawpay.h
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gstrtpvrawpay.h \
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gstrtpstreampay.c
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EXTRA_DIST = dboolhuff.LICENSE
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@ -97,6 +97,7 @@
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#include "gstrtpvp8pay.h"
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#include "gstrtpvrawdepay.h"
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#include "gstrtpvrawpay.h"
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#include "gstrtpstreampay.h"
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static gboolean
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plugin_init (GstPlugin * plugin)
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@ -325,6 +326,9 @@ plugin_init (GstPlugin * plugin)
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if (!gst_rtp_vraw_pay_plugin_init (plugin))
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return FALSE;
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if (!gst_rtp_stream_pay_plugin_init (plugin))
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return FALSE;
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return TRUE;
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}
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287
gst/rtp/gstrtpstreampay.c
Normal file
287
gst/rtp/gstrtpstreampay.c
Normal file
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@ -0,0 +1,287 @@
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/*
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* GStreamer
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* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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:element-rtpstreampay
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*
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* Implements stream payloading of RTP and RTCP packets for connection-oriented
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* transport protocols according to RFC4571.
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678
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* gst-launch-1.0 tcpclientsrc port=5678 host=127.0.0.1 do-timestamp=true ! "application/x-rtp-stream,media=audio,clock-rate=48000,encoding-name=VORBIS" ! rtpstreamdepay ! rtpvorbisdepay ! decodebin ! audioconvert ! audioresample ! autoaudiosink
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* ]|
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* </refsect2>
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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 "gstrtpstreampay.h"
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#define GST_CAT_DEFAULT gst_rtp_stream_pay_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp; "
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"application/x-srtp; application/x-srtcp")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp-stream; application/x-rtcp-stream; "
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"application/x-srtp-stream; application/x-srtcp-stream")
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);
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#define parent_class gst_rtp_stream_pay_parent_class
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G_DEFINE_TYPE (GstRtpStreamPay, gst_rtp_stream_pay, GST_TYPE_ELEMENT);
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static gboolean gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static GstFlowReturn gst_rtp_stream_pay_sink_chain (GstPad * pad,
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GstObject * parent, GstBuffer * inbuf);
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static gboolean gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static void
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gst_rtp_stream_pay_class_init (GstRtpStreamPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_pay_debug, "rtpstreampay", 0,
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"RTP stream payloader");
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Stream Payloading", "Codec/Payloader/Network",
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"Payloads RTP/RTCP packets for streaming protocols according to RFC4571",
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"Sebastian Dröge <sebastian@centricular.com>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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}
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static void
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gst_rtp_stream_pay_init (GstRtpStreamPay * self)
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{
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self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_chain));
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gst_pad_set_event_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_event));
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gst_pad_set_query_function (self->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_query));
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gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
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self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (self->srcpad);
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gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
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}
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static GstCaps *
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gst_rtp_stream_pay_sink_get_caps (GstRtpStreamPay * self, GstCaps * filter)
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{
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GstCaps *peerfilter = NULL, *peercaps, *templ;
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GstCaps *res;
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GstStructure *structure;
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guint i, n;
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if (filter) {
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peerfilter = gst_caps_copy (filter);
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n = gst_caps_get_size (peerfilter);
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for (i = 0; i < n; i++) {
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structure = gst_caps_get_structure (peerfilter, i);
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if (gst_structure_has_name (structure, "application/x-rtp"))
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gst_structure_set_name (structure, "application/x-rtp-stream");
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else if (gst_structure_has_name (structure, "application/x-rtcp"))
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gst_structure_set_name (structure, "application/x-rtcp-stream");
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else if (gst_structure_has_name (structure, "application/x-srtp"))
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gst_structure_set_name (structure, "application/x-srtp-stream");
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else
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gst_structure_set_name (structure, "application/x-srtcp-stream");
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}
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}
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templ = gst_pad_get_pad_template_caps (self->sinkpad);
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peercaps = gst_pad_peer_query_caps (self->srcpad, peerfilter);
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if (peercaps) {
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/* Rename structure names */
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peercaps = gst_caps_make_writable (peercaps);
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n = gst_caps_get_size (peercaps);
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for (i = 0; i < n; i++) {
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structure = gst_caps_get_structure (peercaps, i);
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if (gst_structure_has_name (structure, "application/x-rtp-stream"))
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gst_structure_set_name (structure, "application/x-rtp");
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else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
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gst_structure_set_name (structure, "application/x-rtcp");
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else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
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gst_structure_set_name (structure, "application/x-srtp");
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else
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gst_structure_set_name (structure, "application/x-srtcp");
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}
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res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (peercaps);
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} else {
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res = templ;
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}
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (res);
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res = intersection;
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gst_caps_unref (peerfilter);
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}
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return res;
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}
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static gboolean
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gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
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GstQuery * query)
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{
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GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
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gboolean ret;
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GST_LOG_OBJECT (pad, "Handling query of type '%s'",
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gst_query_type_get_name (GST_QUERY_TYPE (query)));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CAPS:
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{
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GstCaps *caps;
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gst_query_parse_caps (query, &caps);
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caps = gst_rtp_stream_pay_sink_get_caps (self, caps);
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gst_query_set_caps_result (query, caps);
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gst_caps_unref (caps);
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ret = TRUE;
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break;
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}
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default:
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ret = gst_pad_query_default (pad, parent, query);
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}
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return ret;
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}
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static gboolean
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gst_rtp_stream_pay_sink_set_caps (GstRtpStreamPay * self, GstCaps * caps)
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{
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GstCaps *othercaps;
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GstStructure *structure;
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gboolean ret;
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othercaps = gst_caps_copy (caps);
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structure = gst_caps_get_structure (othercaps, 0);
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if (gst_structure_has_name (structure, "application/x-rtp"))
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gst_structure_set_name (structure, "application/x-rtp-stream");
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else if (gst_structure_has_name (structure, "application/x-rtcp"))
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gst_structure_set_name (structure, "application/x-rtcp-stream");
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else if (gst_structure_has_name (structure, "application/x-srtp"))
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gst_structure_set_name (structure, "application/x-srtp-stream");
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else
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gst_structure_set_name (structure, "application/x-srtcp-stream");
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ret = gst_pad_set_caps (self->srcpad, othercaps);
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gst_caps_unref (othercaps);
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return ret;
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}
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static gboolean
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gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event)
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{
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GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
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gboolean ret;
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GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CAPS:
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{
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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ret = gst_rtp_stream_pay_sink_set_caps (self, caps);
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gst_event_unref (event);
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break;
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}
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default:
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ret = gst_pad_event_default (pad, parent, event);
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break;
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_stream_pay_sink_chain (GstPad * pad, GstObject * parent,
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GstBuffer * inbuf)
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{
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GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
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GstBuffer *outbuf;
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gsize size;
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guint8 size16[2];
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size = gst_buffer_get_size (inbuf);
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if (size > G_MAXUINT16) {
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GST_ELEMENT_ERROR (self, CORE, FAILED, (NULL),
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("Only buffers up to %d bytes supported, got %" G_GSIZE_FORMAT,
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G_MAXUINT16, size));
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gst_buffer_unref (inbuf);
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return GST_FLOW_ERROR;
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}
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outbuf = gst_buffer_new_and_alloc (2);
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GST_WRITE_UINT16_BE (size16, size);
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gst_buffer_fill (outbuf, 0, size16, 2);
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gst_buffer_copy_into (outbuf, inbuf, GST_BUFFER_COPY_ALL, 0, -1);
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gst_buffer_unref (inbuf);
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return gst_pad_push (self->srcpad, outbuf);
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}
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gboolean
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gst_rtp_stream_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpstreampay",
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GST_RANK_NONE, GST_TYPE_RTP_STREAM_PAY);
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}
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54
gst/rtp/gstrtpstreampay.h
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54
gst/rtp/gstrtpstreampay.h
Normal file
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@ -0,0 +1,54 @@
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/*
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* GStreamer
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* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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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#ifndef __GST_RTP_STREAM_PAY_H__
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#define __GST_RTP_STREAM_PAY_H__
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#include <gst/gst.h>
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G_BEGIN_DECLS
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#define GST_TYPE_RTP_STREAM_PAY (gst_rtp_stream_pay_get_type())
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#define GST_RTP_STREAM_PAY(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPay))
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#define GST_IS_RTP_STREAM_PAY(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_STREAM_PAY))
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#define GST_RTP_STREAM_PAY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPayClass))
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#define GST_IS_RTP_STREAM_PAY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_RTP_STREAM_PAY))
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#define GST_RTP_STREAM_PAY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPayClass))
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typedef struct _GstRtpStreamPay GstRtpStreamPay;
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typedef struct _GstRtpStreamPayClass GstRtpStreamPayClass;
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struct _GstRtpStreamPay {
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GstElement parent;
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GstPad *srcpad, *sinkpad;
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};
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struct _GstRtpStreamPayClass {
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GstElementClass parent_class;
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};
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GType gst_rtp_stream_pay_get_type (void);
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gboolean gst_rtp_stream_pay_plugin_init (GstPlugin * plugin);
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G_END_DECLS
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#endif /* __GST_RTP_STREAM_PAY_H__ */
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