diff -r 71e347f905f2 -r 4a7fac7dd34a gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c
--- a/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c	Fri Mar 19 09:35:09 2010 +0200
+++ b/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c	Fri Apr 16 15:15:52 2010 +0300
@@ -1,6 +1,6 @@
 /* GStreamer
  *
- * unit test for audioresample
+ * unit test for audioresample, based on the audioresample unit test
  *
  * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
  * Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
@@ -20,27 +20,37 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
-
-
 #include <gst/gst_global.h>
 #include <unistd.h>
+
 #include <gst/check/gstcheck.h>
 
-
-
+#include <gst/audio/audio.h>
 
 #define LOG_FILE "c:\\logs\\audioresample_logs.txt" 
 #include "std_log_result.h" 
 #define LOG_FILENAME_LINE __FILE__, __LINE__
 
+//char* xmlfile = "gstsystemclock";
+
 void create_xml(int result)
 {
+
     if(result)
+    {
         assert_failed = 1;
-    
+    } 
+
     testResultXml(xmlfile);
     close_log_file();
+
+    if(result)
+    {
+        exit (-1);
+    }    
+
 }
+
 #include "libgstreamer_wsd_solution.h" 
 
 
@@ -79,16 +89,19 @@
 #endif
 
 
-
-
-
 /* For ease of programming we use globals to keep refs for our floating
  * src and sink pads we create; otherwise we always have to do get_pad,
  * get_peer, and then remove references in every test function */
 static GstPad *mysrcpad, *mysinkpad;
 
+#define RESAMPLE_CAPS_FLOAT     \
+    "audio/x-raw-float, "               \
+    "channels = (int) [ 1, MAX ], "     \
+    "rate = (int) [ 1,  MAX ], "        \
+    "endianness = (int) BYTE_ORDER, "   \
+    "width = (int) { 32, 64 }"
 
-#define RESAMPLE_CAPS_TEMPLATE_STRING   \
+#define RESAMPLE_CAPS_INT       \
     "audio/x-raw-int, "                 \
     "channels = (int) [ 1, MAX ], "     \
     "rate = (int) [ 1,  MAX ], "        \
@@ -97,6 +110,10 @@
     "depth = (int) 16, "                \
     "signed = (bool) TRUE"
 
+#define RESAMPLE_CAPS_TEMPLATE_STRING   \
+    RESAMPLE_CAPS_FLOAT " ; " \
+    RESAMPLE_CAPS_INT
+
 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
@@ -109,20 +126,25 @@
     );
 
 static GstElement *
-setup_audioresample (int channels, int inrate, int outrate)
+setup_audioresample (int channels, int inrate, int outrate, int width,
+    gboolean fp)
 {
   GstElement *audioresample;
   GstCaps *caps;
   GstStructure *structure;
-  GstPad *pad;
 
   GST_DEBUG ("setup_audioresample");
   audioresample = gst_check_setup_element ("audioresample");
 
-  caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+  if (fp)
+    caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+  else
+    caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
   structure = gst_caps_get_structure (caps, 0);
   gst_structure_set (structure, "channels", G_TYPE_INT, channels,
-      "rate", G_TYPE_INT, inrate, NULL);
+      "rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
+  if (!fp)
+    gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
   fail_unless (gst_caps_is_fixed (caps));
 
   fail_unless (gst_element_set_state (audioresample,
@@ -130,27 +152,30 @@
       "could not set to paused");
 
   mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
-  pad = gst_pad_get_peer (mysrcpad);
-  gst_pad_set_caps (pad, caps);
-  gst_object_unref (GST_OBJECT (pad));
+  gst_pad_set_caps (mysrcpad, caps);
   gst_caps_unref (caps);
-  gst_pad_set_active (mysrcpad, TRUE);
 
-  caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+  if (fp)
+    caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+  else
+    caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
   structure = gst_caps_get_structure (caps, 0);
   gst_structure_set (structure, "channels", G_TYPE_INT, channels,
-      "rate", G_TYPE_INT, outrate, NULL);
+      "rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
+  if (!fp)
+    gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
   fail_unless (gst_caps_is_fixed (caps));
 
   mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
   /* this installs a getcaps func that will always return the caps we set
    * later */
+  gst_pad_set_caps (mysinkpad, caps);
   gst_pad_use_fixed_caps (mysinkpad);
-  pad = gst_pad_get_peer (mysinkpad);
-  gst_pad_set_caps (pad, caps);
-  gst_object_unref (GST_OBJECT (pad));
+
+  gst_pad_set_active (mysinkpad, TRUE);
+  gst_pad_set_active (mysrcpad, TRUE);
+
   gst_caps_unref (caps);
-  gst_pad_set_active (mysinkpad, TRUE);
 
   return audioresample;
 }
@@ -183,8 +208,11 @@
     buffer = GST_BUFFER (l->data);
     ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
     GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
-        G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buffer),
-        GST_BUFFER_DURATION (buffer));
+        G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
+        G_GUINT64_FORMAT,
+        GST_BUFFER_TIMESTAMP (buffer),
+        GST_BUFFER_DURATION (buffer),
+        GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
 
     fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
     fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
@@ -207,12 +235,12 @@
   GstElement *audioresample;
   GstBuffer *inbuffer, *outbuffer;
   GstCaps *caps;
+  guint64 offset = 0;
 
   int i, j;
   gint16 *p;
-  
 
-  audioresample = setup_audioresample (2, inrate, outrate);
+  audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
   caps = gst_pad_get_negotiated_caps (mysrcpad);
   fail_unless (gst_caps_is_fixed (caps));
 
@@ -223,10 +251,11 @@
   for (j = 1; j <= numbuffers; ++j) {
 
     inbuffer = gst_buffer_new_and_alloc (samples * 4);
-    GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
+    GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
     GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
-    GST_BUFFER_OFFSET (inbuffer) = 0;
-    GST_BUFFER_OFFSET_END (inbuffer) = samples;
+    GST_BUFFER_OFFSET (inbuffer) = offset;
+    offset += samples;
+    GST_BUFFER_OFFSET_END (inbuffer) = offset;
 
     gst_buffer_set_caps (inbuffer, caps);
 
@@ -264,9 +293,9 @@
  */
 void test_perfect_stream()
 {
-	xmlfile = "test_perfect_stream";
+  /* integral scalings */
+    xmlfile = "test_perfect_stream";
   std_log(LOG_FILENAME_LINE, "Test Started test_perfect_stream");
-  /* integral scalings */
   test_perfect_stream_instance (48000, 24000, 500, 20);
   test_perfect_stream_instance (48000, 12000, 500, 20);
   test_perfect_stream_instance (12000, 24000, 500, 20);
@@ -299,7 +328,10 @@
   int i, j;
   gint16 *p;
 
-  audioresample = setup_audioresample (2, inrate, outrate);
+  GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
+      inrate, outrate, samples, numbuffers);
+
+  audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
   caps = gst_pad_get_negotiated_caps (mysrcpad);
   fail_unless (gst_caps_is_fixed (caps));
 
@@ -329,6 +361,11 @@
       ++p;
     }
 
+    GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
+        G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+        G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
+        GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
+        GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
     /* pushing gives away my reference ... */
     fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
 
@@ -336,9 +373,14 @@
     outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
     fail_if (outbuffer == NULL);
     fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
+    GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
+        G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+        G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
+        GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
+        GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
     if (j > 1) {
       fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
-          "expected discont buffer");
+          "expected discont for buffer #%d", j);
     }
   }
 
@@ -349,10 +391,9 @@
 
 void test_discont_stream()
 {
-	xmlfile = "test_discont_stream";
+  /* integral scalings */
+    xmlfile = "test_discont_stream";
   std_log(LOG_FILENAME_LINE, "Test Started test_discont_stream");
-
-  /* integral scalings */
   test_discont_stream_instance (48000, 24000, 500, 20);
   test_discont_stream_instance (48000, 12000, 500, 20);
   test_discont_stream_instance (12000, 24000, 500, 20);
@@ -364,7 +405,6 @@
 
   /* wacky scalings */
   test_discont_stream_instance (12345, 54321, 500, 20);
-
   test_discont_stream_instance (101, 99, 500, 20);
   
   std_log(LOG_FILENAME_LINE, "Test Successful");
@@ -382,9 +422,8 @@
   GstBuffer *inbuffer;
   GstCaps *caps;
   xmlfile = "test_reuse";
-  std_log(LOG_FILENAME_LINE, "Test Started test_reuse");
-
-  audioresample = setup_audioresample (1, 9343, 48000);
+std_log(LOG_FILENAME_LINE, "Test Started test_reuse");
+  audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
   caps = gst_pad_get_negotiated_caps (mysrcpad);
   fail_unless (gst_caps_is_fixed (caps));
 
@@ -436,6 +475,7 @@
 
   cleanup_audioresample (audioresample);
   gst_caps_unref (caps);
+  
   std_log(LOG_FILENAME_LINE, "Test Successful");
   create_xml(0);
 }
@@ -448,10 +488,10 @@
   GstCaps *caps;
   guint i;
   xmlfile = "test_shutdown";
-  std_log(LOG_FILENAME_LINE, "Test Started test_shutdown");
-
+std_log(LOG_FILENAME_LINE, "Test Started test_shutdown");
   /* create pipeline, force audioresample to actually resample */
   pipeline = gst_pipeline_new (NULL);
+
   src = gst_check_setup_element ("audiotestsrc");
   cf1 = gst_check_setup_element ("capsfilter");
   ar = gst_check_setup_element ("audioresample");
@@ -486,31 +526,341 @@
   }
 
   gst_object_unref (pipeline);
+  
+  std_log(LOG_FILENAME_LINE, "Test Successful");
+  create_xml(0);
+}
+
+
+
+static GstFlowReturn
+live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
+    guint size, GstCaps * caps, GstBuffer ** buf)
+{
+  GstStructure *structure;
+  gint rate;
+  gint channels;
+  GstCaps *desired;
+
+  structure = gst_caps_get_structure (caps, 0);
+  fail_unless (gst_structure_get_int (structure, "rate", &rate));
+  fail_unless (gst_structure_get_int (structure, "channels", &channels));
+
+  if (rate < 48000)
+    return GST_FLOW_NOT_NEGOTIATED;
+
+  desired = gst_caps_copy (caps);
+  gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
+
+  *buf = gst_buffer_new_and_alloc (channels * 48000);
+  gst_buffer_set_caps (*buf, desired);
+  gst_caps_unref (desired);
+
+  return GST_FLOW_OK;
+}
+
+static GstCaps *
+live_switch_get_sink_caps (GstPad * pad)
+{
+  GstCaps *result;
+
+  result = gst_caps_copy (GST_PAD_CAPS (pad));
+
+  gst_caps_set_simple (result,
+      "rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
+
+  return result;
+}
+
+static void
+live_switch_push (int rate, GstCaps * caps)
+{
+  GstBuffer *inbuffer;
+  GstCaps *desired;
+  GList *l;
+
+  desired = gst_caps_copy (caps);
+  gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
+  gst_pad_set_caps (mysrcpad, desired);
+
+  fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
+          GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
+
+  /* When the basetransform hits the non-configured case it always
+   * returns a buffer with exactly the same caps as we requested so the actual
+   * renegotiation (if needed) will be done in the _chain*/
+  fail_unless (inbuffer != NULL);
+  GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
+      desired, GST_BUFFER_CAPS (inbuffer));
+  fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
+
+  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
+  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
+  GST_BUFFER_OFFSET (inbuffer) = 0;
+
+  /* pushing gives away my reference ... */
+  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+  /* ... but it ends up being collected on the global buffer list */
+  fail_unless_equals_int (g_list_length (buffers), 1);
+
+  for (l = buffers; l; l = l->next) {
+    GstBuffer *buffer = GST_BUFFER (l->data);
+
+    gst_buffer_unref (buffer);
+  }
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_caps_unref (desired);
+}
+
+void test_live_switch()
+{
+  GstElement *audioresample;
+  GstEvent *newseg;
+  GstCaps *caps;
+  xmlfile = "test_live_switch";
+std_log(LOG_FILENAME_LINE, "Test Started test_live_switch");
+  audioresample = setup_audioresample (4, 48000, 48000, 16, FALSE);
+
+  /* Let the sinkpad act like something that can only handle things of
+   * rate 48000- and can only allocate buffers for that rate, but if someone
+   * tries to get a buffer with a rate higher then 48000 tries to renegotiate
+   * */
+  gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
+  gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
+
+  gst_pad_use_fixed_caps (mysrcpad);
+
+  caps = gst_pad_get_negotiated_caps (mysrcpad);
+  fail_unless (gst_caps_is_fixed (caps));
+
+  fail_unless (gst_element_set_state (audioresample,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
+  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
+
+  /* downstream can provide the requested rate, a buffer alloc will be passed
+   * on */
+  live_switch_push (48000, caps);
+
+  /* Downstream can never accept this rate, buffer alloc isn't passed on */
+  live_switch_push (40000, caps);
+
+  /* Downstream can provide the requested rate but will re-negotiate */
+  live_switch_push (50000, caps);
+
+  cleanup_audioresample (audioresample);
+  gst_caps_unref (caps);
+  
   std_log(LOG_FILENAME_LINE, "Test Successful");
   create_xml(0);
 }
 
-/*
-audioresample_suite (void)
+
+
+#ifndef GST_DISABLE_PARSE
+
+static GMainLoop *loop;
+static gint messages = 0;
+
+static void
+element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+  gchar *s;
+
+  s = gst_structure_to_string (gst_message_get_structure (message));
+  GST_DEBUG ("Received message: %s", s);
+  g_free (s);
+
+  messages++;
+}
+
+static void
+eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+  GST_DEBUG ("Received eos");
+  g_main_loop_quit (loop);
+}
+
+static void
+test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
+{
+  GstElement *pipeline;
+  GstBus *bus;
+  GError *error = NULL;
+  gchar *pipe_str;
+
+  pipe_str =
+      g_strdup_printf
+      ("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
+      (fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
+      outrate, width);
+
+  pipeline = gst_parse_launch (pipe_str, &error);
+  fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+      error ? error->message : "(invalid error)");
+  g_free (pipe_str);
+
+  bus = gst_element_get_bus (pipeline);
+  fail_if (bus == NULL);
+  gst_bus_add_signal_watch (bus);
+  g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
+      NULL);
+  g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  /* run until we receive EOS */
+  loop = g_main_loop_new (NULL, FALSE);
+
+  g_main_loop_run (loop);
+
+  g_main_loop_unref (loop);
+  loop = NULL;
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+
+  fail_if (messages > 0, "Received imperfect timestamp messages");
+  gst_object_unref (pipeline);
+}
+
+void test_pipelines()
 {
-test_perfect_stream();
-test_discont_stream();
-test_reuse();
-test_shutdown();
-}*/
+  gint quality;
+  xmlfile = "test_pipelines";
+std_log(LOG_FILENAME_LINE, "Test Started test_pipelines");
+  /* Test qualities 0, 5 and 10 */
+  for (quality = 0; quality < 11; quality += 5) {
+    test_pipeline (8, FALSE, 44100, 48000, quality);
+    test_pipeline (8, FALSE, 48000, 44100, quality);
+
+    test_pipeline (16, FALSE, 44100, 48000, quality);
+    test_pipeline (16, FALSE, 48000, 44100, quality);
+
+    test_pipeline (24, FALSE, 44100, 48000, quality);
+    test_pipeline (24, FALSE, 48000, 44100, quality);
+
+    test_pipeline (32, FALSE, 44100, 48000, quality);
+    test_pipeline (32, FALSE, 48000, 44100, quality);
+
+    test_pipeline (32, TRUE, 44100, 48000, quality);
+    test_pipeline (32, TRUE, 48000, 44100, quality);
+
+    test_pipeline (64, TRUE, 44100, 48000, quality);
+    test_pipeline (64, TRUE, 48000, 44100, quality); 
+  }
+  
+  std_log(LOG_FILENAME_LINE, "Test Successful");
+  create_xml(0);
+}
+
+
+
+void test_preference_passthrough()
+{
+  GstStateChangeReturn ret;
+  GstElement *pipeline, *src;
+  GstStructure *s;
+  GstMessage *msg;
+  GstCaps *caps;
+  GstPad *pad;
+  GstBus *bus;
+  GError *error = NULL;
+  gint rate = 0;
+
+  xmlfile = "test_preference_passthrough";
+std_log(LOG_FILENAME_LINE, "Test Started test_preference_passthrough");
+  pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
+      "audioresample ! "
+      "audio/x-raw-int,rate=8000,channels=1,width=16,depth=16,signed=(boolean)true,endianness=(int)BYTE_ORDER ! "
+      "fakesink can-activate-pull=0 ", &error);
+  fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+      error ? error->message : "(invalid error)");
+
+  ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
+  fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
+
+  /* run until we receive EOS */
+  bus = gst_element_get_bus (pipeline);
+  fail_if (bus == NULL);
+  msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
+  gst_message_unref (msg);
+  gst_object_unref (bus);
+
+  src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
+  fail_unless (src != NULL);
+  pad = gst_element_get_static_pad (src, "src");
+  fail_unless (pad != NULL);
+  caps = gst_pad_get_negotiated_caps (pad);
+  GST_LOG ("negotiated audiotestsrc caps: %" GST_PTR_FORMAT, caps);
+  fail_unless (caps != NULL);
+  s = gst_caps_get_structure (caps, 0);
+  fail_unless (gst_structure_get_int (s, "rate", &rate));
+  /* there's no need to resample, audiotestsrc supports any rate, so make
+   * sure audioresample provided upstream with the right caps to negotiate
+   * this correctly */
+  fail_unless_equals_int (rate, 8000);
+  gst_caps_unref (caps);
+  gst_object_unref (pad);
+  gst_object_unref (src);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+  
+  std_log(LOG_FILENAME_LINE, "Test Successful");
+  create_xml(0);
+}
+
+
+
+#endif
+
+//static Suite *
+//audioresample_suite (void)
+//{
+//  Suite *s = suite_create ("audioresample");
+//  TCase *tc_chain = tcase_create ("general");
+//
+//  suite_add_tcase (s, tc_chain);
+//  tcase_add_test (tc_chain, test_perfect_stream);
+//  tcase_add_test (tc_chain, test_discont_stream);
+//  tcase_add_test (tc_chain, test_reuse);
+//  tcase_add_test (tc_chain, test_shutdown);
+//  tcase_add_test (tc_chain, test_live_switch);
+//
+//#ifndef GST_DISABLE_PARSE
+//  tcase_set_timeout (tc_chain, 360);
+//  tcase_add_test (tc_chain, test_pipelines);
+//  tcase_add_test (tc_chain, test_preference_passthrough);
+//#endif
+//
+//  return s;
+//}
 
 void (*fn[]) (void) = {
 test_perfect_stream,
 test_discont_stream,
 test_reuse,
-test_shutdown
+test_shutdown,
+test_live_switch,
+test_pipelines,
+test_preference_passthrough
 };
 
 char *args[] = {
 "test_perfect_stream",
 "test_discont_stream",
 "test_reuse",
-"test_shutdown"
+"test_shutdown",
+"test_live_switch",
+"test_pipelines",
+"test_preference_passthrough"
 };
 
 GST_CHECK_MAIN (audioresample);