/* GStreamer * Copyright (C) 2013 Sebastian Dröge * * 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 "); 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); }