Add dccp plugin. Fixes #542390.

Original commit message from CVS:
patch by: Leandro Melo de Sales <leandroal@gmail.com>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-dccp.xml:
* gst/dccp/Makefile.am:
* gst/dccp/gstdccp.c:
* gst/dccp/gstdccp.h:
* gst/dccp/gstdccpclientsink.c:
* gst/dccp/gstdccpclientsink.h:
* gst/dccp/gstdccpclientsrc.c:
* gst/dccp/gstdccpclientsrc.h:
* gst/dccp/gstdccpplugin.c:
* gst/dccp/gstdccpserversink.c:
* gst/dccp/gstdccpserversink.h:
* gst/dccp/gstdccpserversrc.c:
* gst/dccp/gstdccpserversrc.h:
* tests/icles/dccp/README:
* tests/icles/dccp/call/README:
* tests/icles/dccp/call/DCCPClient.c:
* tests/icles/dccp/call/DCCPServer.c:
* tests/icles/dccp/file/DCCPClientSaveFile.c:
* tests/icles/dccp/file/DCCPServerSendFile.c:
* tests/icles/dccp/mic/DCCPClientPlayMic.c:
* tests/icles/dccp/mic/DCCPServerMic.c:
* tests/icles/dccp/mp3/DCCPClientPlayMP3.c:
* tests/icles/dccp/mp3/DCCPServerSendMP3.c:
* tests/icles/dccp/mp3Speex/DCCPClientPlaySpeexMP3.c:
* tests/icles/dccp/mp3Speex/DCCPServerSendSpeexMP3.c:
* tests/icles/dccp/mp3Stream/DCCPClientPlayMP3Stream.c:
* tests/icles/dccp/mp3Stream/DCCPServerSendMP3Stream.c:
Add dccp plugin. Fixes #542390.

2 files changed, 248 insertions, 0 deletions

diff --git a/tests/icles/dccp/mp3Speex/DCCPClientPlaySpeexMP3.c b/tests/icles/dccp/mp3Speex/DCCPClientPlaySpeexMP3.c
new file mode 100644
index 00000000..d05b4198
--- /dev/null
+++ b/tests/icles/dccp/mp3Speex/DCCPClientPlaySpeexMP3.c
@@ -0,0 +1,120 @@
+/* GStreamer
+ * Copyright (C) <2007> Leandro Melo de Sales <leandroal@gmail.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <gst/gst.h>
+
+static gboolean
+bus_call (GstBus * bus, GstMessage * msg, gpointer data)
+{
+
+  GMainLoop *loop = (GMainLoop *) data;
+
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_EOS:
+      g_print ("End-of-stream\n");
+      g_main_loop_quit (loop);
+      break;
+    case GST_MESSAGE_ERROR:{
+      gchar *debug;
+      GError *err;
+
+      gst_message_parse_error (msg, &err, &debug);
+      g_free (debug);
+
+      g_print ("Error: %s\n", err->message);
+      g_error_free (err);
+
+      g_main_loop_quit (loop);
+      break;
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+
+  GMainLoop *loop;
+  GstBus *bus;
+  GstElement *pipeline, *alsasink, *rtpspeexdepay, *speexdec, *dccpclientsrc;
+  GstCaps *caps;
+
+  /* initialize GStreamer */
+  gst_init (&argc, &argv);
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* check input arguments */
+  if (argc != 3) {
+    g_print ("%s\n", "see usage: serverHost serverPort");
+    return -1;
+  }
+
+  /* create elements */
+  pipeline = gst_pipeline_new ("audio-sender");
+  alsasink = gst_element_factory_make ("alsasink", "alsa-sink");
+  rtpspeexdepay = gst_element_factory_make ("rtpspeexdepay", "rtpspeexdepay");
+  speexdec = gst_element_factory_make ("speexdec", "speexdec");
+  dccpclientsrc = gst_element_factory_make ("dccpclientsrc", "client-source");
+
+  if (!pipeline || !alsasink || !rtpspeexdepay || !speexdec || !dccpclientsrc) {
+    g_print ("One element could not be created\n");
+    return -1;
+  }
+
+  caps =
+      gst_caps_from_string
+      ("application/x-rtp, media=(string)audio, payload=(int)110, clock-rate=(int)44100, encoding-name=(string)SPEEX, ssrc=(guint)152981653, clock-base=(guint)1553719649, seqnum-base=(guint)3680, encoding-params=(string)1");
+  g_object_set (G_OBJECT (dccpclientsrc), "caps", caps, NULL);
+  gst_object_unref (caps);
+
+  g_object_set (G_OBJECT (dccpclientsrc), "host", argv[1], NULL);
+  g_object_set (G_OBJECT (dccpclientsrc), "port", atoi (argv[2]), NULL);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+  gst_bus_add_watch (bus, bus_call, loop);
+  gst_object_unref (bus);
+
+  /* put all elements in a bin */
+  gst_bin_add_many (GST_BIN (pipeline), dccpclientsrc, rtpspeexdepay, speexdec,
+      alsasink, NULL);
+
+  gst_element_link_many (dccpclientsrc, rtpspeexdepay, speexdec, alsasink,
+      NULL);
+
+  /* Now set to playing and iterate. */
+  g_print ("Setting to PLAYING\n");
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+  g_print ("Running\n");
+  g_main_loop_run (loop);
+
+  /* clean up nicely */
+  g_print ("Returned, stopping playback\n");
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  g_print ("Deleting pipeline\n");
+  gst_object_unref (GST_OBJECT (pipeline));
+
+  return 0;
+}
diff --git a/tests/icles/dccp/mp3Speex/DCCPServerSendSpeexMP3.c b/tests/icles/dccp/mp3Speex/DCCPServerSendSpeexMP3.c
new file mode 100644
index 00000000..58166dd1
--- /dev/null
+++ b/tests/icles/dccp/mp3Speex/DCCPServerSendSpeexMP3.c
@@ -0,0 +1,128 @@
+/* GStreamer
+ * Copyright (C) <2007> Leandro Melo de Sales <leandroal@gmail.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <gst/gst.h>
+
+static gboolean
+bus_call (GstBus * bus, GstMessage * msg, gpointer data)
+{
+
+  GMainLoop *loop = (GMainLoop *) data;
+
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_EOS:
+      g_print ("End-of-stream\n");
+      g_main_loop_quit (loop);
+      break;
+    case GST_MESSAGE_ERROR:{
+      gchar *debug;
+      GError *err;
+
+      gst_message_parse_error (msg, &err, &debug);
+      g_free (debug);
+
+      g_print ("Error: %s\n", err->message);
+      g_error_free (err);
+
+      g_main_loop_quit (loop);
+      break;
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+
+  GMainLoop *loop;
+  GstBus *bus;
+  GstElement *pipeline, *filesrc, *mad, *audioconvert, *capsfilter, *speexenc,
+      *rtpspeexpay, *dccpserversink;
+  GstCaps *caps;
+
+  /* initialize GStreamer */
+  gst_init (&argc, &argv);
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* check input arguments */
+  if (argc != 3) {
+    g_print ("%s\n", "see usage: port mp3Location");
+    return -1;
+  }
+
+  /* create elements */
+  pipeline = gst_pipeline_new ("audio-sender");
+  filesrc = gst_element_factory_make ("filesrc", "file-source");
+  mad = gst_element_factory_make ("mad", "mad");
+  audioconvert = gst_element_factory_make ("audioconvert", "audioconvert");
+  capsfilter = gst_element_factory_make ("capsfilter", "capsfilter");
+  speexenc = gst_element_factory_make ("speexenc", "speexenc");
+  rtpspeexpay = gst_element_factory_make ("rtpspeexpay", "rtpspeexpay");
+  dccpserversink = gst_element_factory_make ("dccpserversink", "server-sink");
+
+
+  if (!pipeline || !filesrc || !dccpserversink || !mad || !audioconvert
+      || !capsfilter || !speexenc || !rtpspeexpay) {
+    g_print ("One element could not be created\n");
+    return -1;
+  }
+
+  g_object_set (G_OBJECT (dccpserversink), "port", atoi (argv[1]), NULL);
+  g_object_set (G_OBJECT (filesrc), "location", argv[2], NULL);
+
+  caps =
+      gst_caps_from_string
+      ("audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)1");
+  g_object_set (G_OBJECT (capsfilter), "caps", caps, NULL);
+  gst_object_unref (caps);
+
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+  gst_bus_add_watch (bus, bus_call, loop);
+  gst_object_unref (bus);
+
+  /* put all elements in a bin */
+  gst_bin_add_many (GST_BIN (pipeline), filesrc, mad, audioconvert, capsfilter,
+      speexenc, rtpspeexpay, dccpserversink, NULL);
+
+  gst_element_link_many (filesrc, mad, audioconvert, capsfilter, speexenc,
+      rtpspeexpay, dccpserversink, NULL);
+
+
+  /* Now set to playing and iterate. */
+  g_print ("Setting to PLAYING\n");
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+  g_print ("Running\n");
+  g_main_loop_run (loop);
+
+  /* clean up nicely */
+  g_print ("Returned, stopping playback\n");
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  g_print ("Deleting pipeline\n");
+  gst_object_unref (GST_OBJECT (pipeline));
+
+  return 0;
+}