1 files changed, 120 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;
+}