--- /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-rtpstreamdepay
+ *
+ * Implements stream depayloading 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 "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);
+
+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 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_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));
+
+ 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);
+}
+
+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);
+}
+
+gboolean
+gst_rtp_stream_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpstreamdepay",
+ GST_RANK_NONE, GST_TYPE_RTP_STREAM_DEPAY);
+}
--- /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_DEPAY_H__
+#define __GST_RTP_STREAM_DEPAY_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbaseparse.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_STREAM_DEPAY \
+ (gst_rtp_stream_depay_get_type())
+#define GST_RTP_STREAM_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_STREAM_DEPAY,GstRtpStreamDepay))
+#define GST_RTP_STREAM_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_STREAM_DEPAY,GstRtpStreamDepayClass))
+#define GST_IS_RTP_STREAM_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_STREAM_DEPAY))
+#define GST_IS_RTP_STREAM_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_STREAM_DEPAY))
+
+typedef struct _GstRtpStreamDepay GstRtpStreamDepay;
+typedef struct _GstRtpStreamDepayClass GstRtpStreamDepayClass;
+
+struct _GstRtpStreamDepay
+{
+ GstBaseParse parent;
+};
+
+struct _GstRtpStreamDepayClass
+{
+ GstBaseParseClass parent_class;
+};
+
+GType gst_rtp_stream_depay_get_type (void);
+gboolean gst_rtp_stream_depay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif