/* * 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-rtpstreampay * @title: rtpstreampay * * Implements stream payloading 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 "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 "); gst_element_class_add_static_pad_template (gstelement_class, &src_template); gst_element_class_add_static_pad_template (gstelement_class, &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); }