rtpstreampay: Add RFC4571 RTP stream payloading element

https://bugzilla.gnome.org/show_bug.cgi?id=719829

gst/rtp/Makefile.am

@@ -78,7 +78,8 @@ libgstrtp_la_SOURCES = \
 	gstrtpvp8depay.c \
 	gstrtpvp8pay.c \
 	gstrtpvrawdepay.c  \
-	gstrtpvrawpay.c 
+	gstrtpvrawpay.c \
+	gstrtpstreampay.c
 
 libgstrtp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
 	$(GST_CFLAGS) -Dvp8_norm=gst_rtpvp8_vp8_norm \
@@ -173,7 +174,8 @@ noinst_HEADERS =			\
 	gstrtpvp8depay.h \
 	gstrtpvp8pay.h \
 	gstrtpvrawdepay.h \
-	gstrtpvrawpay.h
+	gstrtpvrawpay.h \
+	gstrtpstreampay.c
 
 EXTRA_DIST = dboolhuff.LICENSE
 

gst/rtp/gstrtp.c

@@ -97,6 +97,7 @@
 #include "gstrtpvp8pay.h"
 #include "gstrtpvrawdepay.h"
 #include "gstrtpvrawpay.h"
+#include "gstrtpstreampay.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
@@ -325,6 +326,9 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtp_vraw_pay_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtp_stream_pay_plugin_init (plugin))
+    return FALSE;
+
   return TRUE;
 }
 

gst/rtp/gstrtpstreampay.c

@@ -0,0 +1,287 @@
+/*
+ * 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);
+}

gst/rtp/gstrtpstreampay.h

@@ -0,0 +1,54 @@
+/* 
+ * 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__ */