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