--- /dev/null
+/*
+ * 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);
+}
--- /dev/null
+/*
+ * 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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