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/* 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-rtpstreamdepay
* @title: rtpstreamdepay
*
* Implements stream depayloading of RTP and RTCP packets for connection-oriented
* transport protocols according to RFC4571.
*
* ## Example launch line
* |[
* 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
* ]|
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpelements.h"
#include "gstrtpstreamdepay.h"
GST_DEBUG_CATEGORY (gst_rtp_stream_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_stream_depay_debug
static GstStaticPadTemplate src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp;"
"application/x-srtp; application/x-srtcp")
);
static GstStaticPadTemplate sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK,
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_depay_parent_class
G_DEFINE_TYPE (GstRtpStreamDepay, gst_rtp_stream_depay, GST_TYPE_BASE_PARSE);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpstreamdepay, "rtpstreamdepay",
GST_RANK_NONE, GST_TYPE_RTP_STREAM_DEPAY, rtp_element_init (plugin));
static gboolean gst_rtp_stream_depay_set_sink_caps (GstBaseParse * parse,
GstCaps * caps);
static GstCaps *gst_rtp_stream_depay_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
static GstFlowReturn gst_rtp_stream_depay_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static gboolean gst_rtp_stream_depay_sink_activate (GstPad * pad,
GstObject * parent);
static void
gst_rtp_stream_depay_class_init (GstRtpStreamDepayClass * klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_depay_debug, "rtpstreamdepay", 0,
"RTP stream depayloader");
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP Stream Depayloading", "Codec/Depayloader/Network",
"Depayloads RTP/RTCP packets for streaming protocols according to RFC4571",
"Sebastian Dröge <sebastian@centricular.com>");
parse_class->set_sink_caps =
GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_set_sink_caps);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_get_sink_caps);
parse_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_handle_frame);
}
static void
gst_rtp_stream_depay_init (GstRtpStreamDepay * self)
{
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (self), 2);
/* Force activation in push mode. We need to get a caps event from upstream
* to know the full RTP caps. */
gst_pad_set_activate_function (GST_BASE_PARSE_SINK_PAD (self),
gst_rtp_stream_depay_sink_activate);
}
static gboolean
gst_rtp_stream_depay_set_sink_caps (GstBaseParse * parse, 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-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");
ret = gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), othercaps);
gst_caps_unref (othercaps);
return ret;
}
static GstCaps *
gst_rtp_stream_depay_get_sink_caps (GstBaseParse * parse, 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-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");
}
}
templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
peercaps =
gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), 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"))
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");
}
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 GstFlowReturn
gst_rtp_stream_depay_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
gsize buf_size;
guint16 size;
if (gst_buffer_extract (frame->buffer, 0, &size, 2) != 2)
return GST_FLOW_ERROR;
size = GUINT16_FROM_BE (size);
buf_size = gst_buffer_get_size (frame->buffer);
/* Need more data */
if (size + 2 > buf_size)
return GST_FLOW_OK;
frame->out_buffer =
gst_buffer_copy_region (frame->buffer, GST_BUFFER_COPY_ALL, 2, size);
return gst_base_parse_finish_frame (parse, frame, size + 2);
}
static gboolean
gst_rtp_stream_depay_sink_activate (GstPad * pad, GstObject * parent)
{
return gst_pad_activate_mode (pad, GST_PAD_MODE_PUSH, TRUE);
}
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