Commit c5284dc0 authored by Sebastian Dröge's avatar Sebastian Dröge

rtpstreampay: Add RFC4571 RTP stream payloading element

https://bugzilla.gnome.org/show_bug.cgi?id=719829
parent 1fd094d9
......@@ -78,7 +78,8 @@ libgstrtp_la_SOURCES = \
gstrtpvp8depay.c \
gstrtpvp8pay.c \
gstrtpvrawdepay.c \
gstrtpvrawpay.c
gstrtpvrawpay.c \
gstrtpstreampay.c
libgstrtp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
$(GST_CFLAGS) -Dvp8_norm=gst_rtpvp8_vp8_norm \
......@@ -173,7 +174,8 @@ noinst_HEADERS = \
gstrtpvp8depay.h \
gstrtpvp8pay.h \
gstrtpvrawdepay.h \
gstrtpvrawpay.h
gstrtpvrawpay.h \
gstrtpstreampay.c
EXTRA_DIST = dboolhuff.LICENSE
......
......@@ -97,6 +97,7 @@
#include "gstrtpvp8pay.h"
#include "gstrtpvrawdepay.h"
#include "gstrtpvrawpay.h"
#include "gstrtpstreampay.h"
static gboolean
plugin_init (GstPlugin * plugin)
......@@ -325,6 +326,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_vraw_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_stream_pay_plugin_init (plugin))
return FALSE;
return TRUE;
}
......
/*
* GStreamer
* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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:element-rtpstreampay
*
* Implements stream payloading of RTP and RTCP packets for connection-oriented
* transport protocols according to RFC4571.
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678
* 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
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpstreampay.h"
#define GST_CAT_DEFAULT gst_rtp_stream_pay_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp; "
"application/x-srtp; application/x-srtcp")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp-stream; application/x-rtcp-stream; "
"application/x-srtp-stream; application/x-srtcp-stream")
);
#define parent_class gst_rtp_stream_pay_parent_class
G_DEFINE_TYPE (GstRtpStreamPay, gst_rtp_stream_pay, GST_TYPE_ELEMENT);
static gboolean gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstFlowReturn gst_rtp_stream_pay_sink_chain (GstPad * pad,
GstObject * parent, GstBuffer * inbuf);
static gboolean gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static void
gst_rtp_stream_pay_class_init (GstRtpStreamPayClass * klass)
{
GstElementClass *gstelement_class;
GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_pay_debug, "rtpstreampay", 0,
"RTP stream payloader");
gstelement_class = (GstElementClass *) klass;
gst_element_class_set_static_metadata (gstelement_class,
"RTP Stream Payloading", "Codec/Payloader/Network",
"Payloads RTP/RTCP packets for streaming protocols according to RFC4571",
"Sebastian Dröge <sebastian@centricular.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
}
static void
gst_rtp_stream_pay_init (GstRtpStreamPay * self)
{
self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_chain));
gst_pad_set_event_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_event));
gst_pad_set_query_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_query));
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (self->srcpad);
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
}
static GstCaps *
gst_rtp_stream_pay_sink_get_caps (GstRtpStreamPay * self, GstCaps * filter)
{
GstCaps *peerfilter = NULL, *peercaps, *templ;
GstCaps *res;
GstStructure *structure;
guint i, n;
if (filter) {
peerfilter = gst_caps_copy (filter);
n = gst_caps_get_size (peerfilter);
for (i = 0; i < n; i++) {
structure = gst_caps_get_structure (peerfilter, i);
if (gst_structure_has_name (structure, "application/x-rtp"))
gst_structure_set_name (structure, "application/x-rtp-stream");
else if (gst_structure_has_name (structure, "application/x-rtcp"))
gst_structure_set_name (structure, "application/x-rtcp-stream");
else if (gst_structure_has_name (structure, "application/x-srtp"))
gst_structure_set_name (structure, "application/x-srtp-stream");
else
gst_structure_set_name (structure, "application/x-srtcp-stream");
}
}
templ = gst_pad_get_pad_template_caps (self->sinkpad);
peercaps = gst_pad_peer_query_caps (self->srcpad, peerfilter);
if (peercaps) {
/* Rename structure names */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
structure = gst_caps_get_structure (peercaps, i);
if (gst_structure_has_name (structure, "application/x-rtp-stream"))
gst_structure_set_name (structure, "application/x-rtp");
else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
gst_structure_set_name (structure, "application/x-rtcp");
else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
gst_structure_set_name (structure, "application/x-srtp");
else
gst_structure_set_name (structure, "application/x-srtcp");
}
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res = templ;
}
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (res);
res = intersection;
gst_caps_unref (peerfilter);
}
return res;
}
static gboolean
gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
gboolean ret;
GST_LOG_OBJECT (pad, "Handling query of type '%s'",
gst_query_type_get_name (GST_QUERY_TYPE (query)));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *caps;
gst_query_parse_caps (query, &caps);
caps = gst_rtp_stream_pay_sink_get_caps (self, caps);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
ret = TRUE;
break;
}
default:
ret = gst_pad_query_default (pad, parent, query);
}
return ret;
}
static gboolean
gst_rtp_stream_pay_sink_set_caps (GstRtpStreamPay * self, GstCaps * caps)
{
GstCaps *othercaps;
GstStructure *structure;
gboolean ret;
othercaps = gst_caps_copy (caps);
structure = gst_caps_get_structure (othercaps, 0);
if (gst_structure_has_name (structure, "application/x-rtp"))
gst_structure_set_name (structure, "application/x-rtp-stream");
else if (gst_structure_has_name (structure, "application/x-rtcp"))
gst_structure_set_name (structure, "application/x-rtcp-stream");
else if (gst_structure_has_name (structure, "application/x-srtp"))
gst_structure_set_name (structure, "application/x-srtp-stream");
else
gst_structure_set_name (structure, "application/x-srtcp-stream");
ret = gst_pad_set_caps (self->srcpad, othercaps);
gst_caps_unref (othercaps);
return ret;
}
static gboolean
gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
gboolean ret;
GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_rtp_stream_pay_sink_set_caps (self, caps);
gst_event_unref (event);
break;
}
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static GstFlowReturn
gst_rtp_stream_pay_sink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
GstBuffer *outbuf;
gsize size;
guint8 size16[2];
size = gst_buffer_get_size (inbuf);
if (size > G_MAXUINT16) {
GST_ELEMENT_ERROR (self, CORE, FAILED, (NULL),
("Only buffers up to %d bytes supported, got %" G_GSIZE_FORMAT,
G_MAXUINT16, size));
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
outbuf = gst_buffer_new_and_alloc (2);
GST_WRITE_UINT16_BE (size16, size);
gst_buffer_fill (outbuf, 0, size16, 2);
gst_buffer_copy_into (outbuf, inbuf, GST_BUFFER_COPY_ALL, 0, -1);
gst_buffer_unref (inbuf);
return gst_pad_push (self->srcpad, outbuf);
}
gboolean
gst_rtp_stream_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpstreampay",
GST_RANK_NONE, GST_TYPE_RTP_STREAM_PAY);
}
/*
* GStreamer
* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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_STREAM_PAY_H__
#define __GST_RTP_STREAM_PAY_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_STREAM_PAY (gst_rtp_stream_pay_get_type())
#define GST_RTP_STREAM_PAY(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPay))
#define GST_IS_RTP_STREAM_PAY(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_STREAM_PAY))
#define GST_RTP_STREAM_PAY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPayClass))
#define GST_IS_RTP_STREAM_PAY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_RTP_STREAM_PAY))
#define GST_RTP_STREAM_PAY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_RTP_STREAM_PAY,GstRtpStreamPayClass))
typedef struct _GstRtpStreamPay GstRtpStreamPay;
typedef struct _GstRtpStreamPayClass GstRtpStreamPayClass;
struct _GstRtpStreamPay {
GstElement parent;
GstPad *srcpad, *sinkpad;
};
struct _GstRtpStreamPayClass {
GstElementClass parent_class;
};
GType gst_rtp_stream_pay_get_type (void);
gboolean gst_rtp_stream_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_STREAM_PAY_H__ */
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