tests/check/: Add add testsuite for the rtp-payloader that tries simulating dataflow. Needs more test data.

Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rtp-payloading.c:
Add add testsuite for the rtp-payloader that tries simulating
dataflow. Needs more test data.

ChangeLog

@@ -1,3 +1,11 @@
+2008-01-29  Stefan Kost  <ensonic@users.sf.net>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.cvsignore:
+	* tests/check/elements/rtp-payloading.c:
+	  Add add testsuite for the rtp-payloader that tries simulating
+	  dataflow. Needs more test data.
+
 2008-01-29  Stefan Kost  <ensonic@users.sf.net>
 
 	* tests/check/elements/alphacolor.c:

tests/check/Makefile.am

@@ -61,10 +61,11 @@ check_PROGRAMS = \
 	elements/audiodynamic \
 	elements/avimux \
 	elements/avisubtitle \
+	elements/icydemux \
 	elements/id3demux \
 	elements/level \
 	elements/matroskamux \
-	elements/icydemux \
+	elements/rtp-payloading \
 	elements/videocrop \
 	elements/videofilter \
 	pipelines/simple-launch-lines \

tests/check/elements/.gitignore

@@ -1,25 +1,26 @@
 .dirstamp
 alphacolor
+apev2mux
 audioamplify
-audiodynamic
-audioinvert
 audiochebyshevfreqband
 audiochebyshevfreqlimit
+audiodynamic
+audioinvert
+audiopanorama
+autodetect
+avimux
 avisubtitle
-level
-matroskamux
 cmmldec
 cmmlenc
 icydemux
-avimux
 id3demux
 id3v2mux
-apev2mux
+level
-audiopanorama
+matroskamux
-videofilter
+rtp-payloading
-videocrop
 sunaudio
-autodetect
+videocrop
+videofilter
 wavpackdec
 wavpackenc
 wavpackparse

tests/check/elements/rtp-payloading.c

@@ -0,0 +1,548 @@
+/* GStreamer
+ * Copyright (C) 2008 Nokia Corporation and its subsidary(-ies)
+ *               contact: <stefan.kost@nokia.com>
+ *
+ * rtp-full.c: Unit test for dataflow in rtppayloaders
+ *
+ * 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 <gst/check/gstcheck.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#define RELEASE_ELEMENT(x) if(x) {gst_object_unref(x); x = NULL;}
+
+#define LOOP_COUNT 1
+
+/*
+ * RTP pipeline structure to store the required elements.
+ */
+typedef struct
+{
+  GstElement *pipeline;
+  GstElement *fdsrc;
+  GstElement *capsfilter;
+  GstElement *rtppay;
+  GstElement *rtpdepay;
+  GstElement *fakesink;
+  int fd[2];
+  const char *frame_data;
+  int frame_data_size;
+  int frame_count;
+} rtp_pipeline;
+
+/*
+ * RTP bus callback.
+ */
+gboolean
+rtp_bus_callback (GstBus * bus, GstMessage * message, gpointer data)
+{
+  GMainLoop *mainloop = (GMainLoop *) data;
+
+  switch (GST_MESSAGE_TYPE (message)) {
+    case GST_MESSAGE_ERROR:
+    {
+      GError *err;
+      gchar *debug;
+
+      gst_message_parse_error (message, &err, &debug);
+      /* FIXME: should we fail the test here */
+      g_print ("Error: %s\n", err->message);
+      g_error_free (err);
+      g_free (debug);
+
+      g_main_loop_quit (mainloop);
+    }
+      break;
+
+    case GST_MESSAGE_EOS:
+    {
+      g_main_loop_quit (mainloop);
+    }
+      break;
+      break;
+
+    default:
+    {
+    }
+      break;
+  }
+
+  return TRUE;
+}
+
+/*
+ * Creates a RTP pipeline for one test.
+ * @param frame_data Pointer to the frame data which is used to pass thru pay/depayloaders.
+ * @param frame_data_size Frame data size in bytes.
+ * @param frame_count Frame count.
+ * @param filtercaps Caps filters.
+ * @param pay Payloader name.
+ * @param depay Depayloader name.
+ * @return
+ * Returns pointer to the RTP pipeline.
+ * The user must free the RTP pipeline when it's not used anymore.
+ */
+rtp_pipeline *
+rtp_pipeline_create (const char *frame_data, int frame_data_size,
+    int frame_count, const char *filtercaps, const char *pay, const char *depay)
+{
+  /* Check parameters. */
+  if (!frame_data || !pay || !depay) {
+    return NULL;
+  }
+
+  /* Allocate memory for the RTP pipeline. */
+  rtp_pipeline *p = (rtp_pipeline *) malloc (sizeof (rtp_pipeline));
+
+  p->frame_data = frame_data;
+  p->frame_data_size = frame_data_size;
+  p->frame_count = frame_count;
+
+  /* Create elements. */
+  p->pipeline = gst_pipeline_new ("rtp_pipeline");
+  p->fdsrc = gst_element_factory_make ("fdsrc", NULL);
+  p->capsfilter = gst_element_factory_make ("capsfilter", NULL);
+  p->rtppay = gst_element_factory_make (pay, NULL);
+  p->rtpdepay = gst_element_factory_make (depay, NULL);
+  p->fakesink = gst_element_factory_make ("fakesink", NULL);
+
+  /* One or more elements are not created successfully or failed to create p? */
+  if (!p->pipeline || !p->fdsrc || !p->capsfilter || !p->rtppay || !p->rtpdepay
+      || !p->fakesink || pipe (p->fd) == -1) {
+    /* Release created elements. */
+    RELEASE_ELEMENT (p->pipeline);
+    RELEASE_ELEMENT (p->fdsrc);
+    RELEASE_ELEMENT (p->capsfilter);
+    RELEASE_ELEMENT (p->rtppay);
+    RELEASE_ELEMENT (p->rtpdepay);
+    RELEASE_ELEMENT (p->fakesink);
+
+    /* Close pipe. */
+    if (p->fd[0]) {
+      close (p->fd[0]);
+    }
+
+    if (p->fd[1]) {
+      close (p->fd[1]);
+    }
+
+    /* Release allocated memory. */
+    free (p);
+
+    return NULL;
+  }
+
+  /* Set fdsrc properties. */
+  g_object_set (p->fdsrc, "fd", p->fd[0], NULL);
+  g_object_set (p->fdsrc, "do-timestamp", TRUE, NULL);
+  g_object_set (p->fdsrc, "blocksize", p->frame_data_size, NULL);
+  g_object_set (p->fdsrc, "num-buffers", p->frame_count * LOOP_COUNT, NULL);
+
+  /* Set caps filters. */
+  GstCaps *caps = gst_caps_from_string (filtercaps);
+
+  g_object_set (p->capsfilter, "caps", caps, NULL);
+  gst_caps_unref (caps);
+
+  /* Add elements to the pipeline. */
+  gst_bin_add (GST_BIN (p->pipeline), p->fdsrc);
+  gst_bin_add (GST_BIN (p->pipeline), p->capsfilter);
+  gst_bin_add (GST_BIN (p->pipeline), p->rtppay);
+  gst_bin_add (GST_BIN (p->pipeline), p->rtpdepay);
+  gst_bin_add (GST_BIN (p->pipeline), p->fakesink);
+
+  /* Link elements. */
+  gst_element_link (p->fdsrc, p->capsfilter);
+  gst_element_link (p->capsfilter, p->rtppay);
+  gst_element_link (p->rtppay, p->rtpdepay);
+  gst_element_link (p->rtpdepay, p->fakesink);
+
+  return p;
+}
+
+/*
+ * Destroys the RTP pipeline.
+ * @param p Pointer to the RTP pipeline.
+ */
+void
+rtp_pipeline_destroy (rtp_pipeline * p)
+{
+  /* Check parameters. */
+  if (p == NULL) {
+    return;
+  }
+
+  /* Release pipeline. */
+  RELEASE_ELEMENT (p->pipeline);
+
+  /* Close pipe. */
+  if (p->fd[0]) {
+    close (p->fd[0]);
+  }
+
+  if (p->fd[1]) {
+    close (p->fd[1]);
+  }
+
+  /* Release allocated memory. */
+  free (p);
+}
+
+/*
+ * Runs the RTP pipeline.
+ * @param p Pointer to the RTP pipeline.
+ */
+void
+rtp_pipeline_run (rtp_pipeline * p)
+{
+  GMainLoop *mainloop = NULL;
+
+  /* Check parameters. */
+  if (p == NULL) {
+    return;
+  }
+
+  /* Create mainloop. */
+  mainloop = g_main_loop_new (NULL, FALSE);
+  if (!mainloop) {
+    return;
+  }
+
+  /* Add bus callback. */
+  GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (p->pipeline));
+
+  gst_bus_add_watch (bus, rtp_bus_callback, (gpointer) mainloop);
+  gst_object_unref (bus);
+
+  /* Set pipeline to PLAYING. */
+  gst_element_set_state (p->pipeline, GST_STATE_PLAYING);
+
+  /* TODO: Writing may need some changes... */
+
+  int i = 0;
+
+  for (; i < LOOP_COUNT; i++) {
+    const char *frame_data_pointer = p->frame_data;
+    int frame_count = p->frame_count;
+
+    /* Write in to the pipe. */
+    while (frame_count > 0) {
+      write (p->fd[1], frame_data_pointer, p->frame_data_size);
+      frame_data_pointer += p->frame_data_size;
+      frame_count--;
+    }
+  }
+
+  /* Run mainloop. */
+  g_main_loop_run (mainloop);
+
+  /* Set pipeline to NULL. */
+  gst_element_set_state (p->pipeline, GST_STATE_NULL);
+
+  /* Release mainloop. */
+  g_main_loop_unref (mainloop);
+}
+
+/*
+ * Creates the RTP pipeline and runs the test using the pipeline.
+ * @param frame_data Pointer to the frame data which is used to pass thru pay/depayloaders.
+ * @param frame_data_size Frame data size in bytes.
+ * @param frame_count Frame count.
+ * @param filtercaps Caps filters.
+ * @param pay Payloader name.
+ * @param depay Depayloader name.
+ */
+void
+rtp_pipeline_test (const char *frame_data, int frame_data_size, int frame_count,
+    const char *filtercaps, const char *pay, const char *depay)
+{
+  /* Create RTP pipeline. */
+  rtp_pipeline *p =
+      rtp_pipeline_create (frame_data, frame_data_size, frame_count, filtercaps,
+      pay, depay);
+  if (p == NULL) {
+    return;
+  }
+
+  /* Run RTP pipeline. */
+  rtp_pipeline_run (p);
+
+  /* Destroy RTP pipeline. */
+  rtp_pipeline_destroy (p);
+}
+
+static char rtp_ilbc_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_ilbc_frame_data_size = 20;
+static int rtp_ilbc_frame_count = 1;
+
+GST_START_TEST (rtp_ilbc)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_ilbc_frame_data, rtp_ilbc_frame_data_size,
+      rtp_ilbc_frame_count, "audio/x-iLBC,mode=20", "rtpilbcpay",
+      "rtpilbcdepay");
+}
+GST_END_TEST
+    static char rtp_gsm_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_gsm_frame_data_size = 20;
+static int rtp_gsm_frame_count = 1;
+
+GST_START_TEST (rtp_gsm)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_gsm_frame_data, rtp_gsm_frame_data_size,
+      rtp_gsm_frame_count, "audio/x-gsm,rate=8000,channels=1", "rtpgsmpay",
+      "rtpgsmdepay");
+}
+GST_END_TEST
+    static char rtp_amr_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_amr_frame_data_size = 20;
+static int rtp_amr_frame_count = 1;
+
+GST_START_TEST (rtp_amr)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_amr_frame_data, rtp_amr_frame_data_size,
+      rtp_amr_frame_count, "audio/AMR,channels=1,rate=8000", "rtpamrpay",
+      "rtpamrdepay");
+}
+GST_END_TEST
+    static char rtp_pcma_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_pcma_frame_data_size = 20;
+static int rtp_pcma_frame_count = 1;
+
+GST_START_TEST (rtp_pcma)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_pcma_frame_data, rtp_pcma_frame_data_size,
+      rtp_pcma_frame_count, "audio/x-alaw,channels=1,rate=8000", "rtppcmapay",
+      "rtppcmadepay");
+}
+GST_END_TEST
+    static char rtp_pcmu_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_pcmu_frame_data_size = 20;
+static int rtp_pcmu_frame_count = 1;
+
+GST_START_TEST (rtp_pcmu)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_pcmu_frame_data, rtp_pcmu_frame_data_size,
+      rtp_pcmu_frame_count, "audio/x-mulaw,channels=1,rate=8000", "rtppcmupay",
+      "rtppcmudepay");
+}
+GST_END_TEST
+    static char rtp_mpa_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_mpa_frame_data_size = 20;
+static int rtp_mpa_frame_count = 1;
+
+GST_START_TEST (rtp_mpa)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_mpa_frame_data, rtp_mpa_frame_data_size,
+      rtp_mpa_frame_count, "audio/mpeg", "rtpmpapay", "rtpmpadepay");
+}
+GST_END_TEST
+    static char rtp_h263_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_h263_frame_data_size = 20;
+static int rtp_h263_frame_count = 1;
+
+GST_START_TEST (rtp_h263)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_h263_frame_data, rtp_h263_frame_data_size,
+      rtp_h263_frame_count, "video/x-h263,variant=itu,h263version=h263",
+      "rtph263pay", "rtph263depay");
+}
+GST_END_TEST
+    static char rtp_h263p_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_h263p_frame_data_size = 20;
+static int rtp_h263p_frame_count = 1;
+
+GST_START_TEST (rtp_h263p)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_h263p_frame_data, rtp_h263p_frame_data_size,
+      rtp_h263p_frame_count, "video/x-h263,variant=itu", "rtph263ppay",
+      "rtph263pdepay");
+}
+GST_END_TEST
+    static char rtp_h264_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_h264_frame_data_size = 20;
+static int rtp_h264_frame_count = 1;
+
+GST_START_TEST (rtp_h264)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_h264_frame_data, rtp_h264_frame_data_size,
+      rtp_h264_frame_count, "video/x-h264", "rtph264pay", "rtph264depay");
+}
+GST_END_TEST
+    static char rtp_L16_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_L16_frame_data_size = 20;
+static int rtp_L16_frame_count = 1;
+
+GST_START_TEST (rtp_L16)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_L16_frame_data, rtp_L16_frame_data_size,
+      rtp_L16_frame_count,
+      "audio/x-raw-int,endianess=4321,signed=true,width=16,depth=16,rate=1,channels=1",
+      "rtpL16pay", "rtpL16depay");
+}
+GST_END_TEST
+    static char rtp_mp2t_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_mp2t_frame_data_size = 20;
+static int rtp_mp2t_frame_count = 1;
+
+GST_START_TEST (rtp_mp2t)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_mp2t_frame_data, rtp_mp2t_frame_data_size,
+      rtp_mp2t_frame_count, "video/mpegts,packetsize=188,systemstream=true",
+      "rtpmp2tpay", "rtpmp2tdepay");
+}
+GST_END_TEST
+    static char rtp_mp4v_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_mp4v_frame_data_size = 20;
+static int rtp_mp4v_frame_count = 1;
+
+GST_START_TEST (rtp_mp4v)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_mp4v_frame_data, rtp_mp4v_frame_data_size,
+      rtp_mp4v_frame_count, "video/mpeg,mpegversion=4,systemstream=false",
+      "rtpmp4vpay", "rtpmp4vdepay");
+}
+GST_END_TEST
+    static char rtp_mp4g_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_mp4g_frame_data_size = 20;
+static int rtp_mp4g_frame_count = 1;
+
+GST_START_TEST (rtp_mp4g)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_mp4g_frame_data, rtp_mp4g_frame_data_size,
+      rtp_mp4g_frame_count, "video/mpeg,mpegversion=4", "rtpmp4gpay",
+      "rtpmp4gdepay");
+}
+GST_END_TEST
+    static char rtp_theora_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_theora_frame_data_size = 20;
+static int rtp_theora_frame_count = 1;
+
+GST_START_TEST (rtp_theora)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_theora_frame_data, rtp_theora_frame_data_size,
+      rtp_theora_frame_count, "video/x-theora", "rtptheorapay",
+      "rtptheoradepay");
+}
+GST_END_TEST
+    static char rtp_vorbis_frame_data[] =
+    { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 };
+static int rtp_vorbis_frame_data_size = 20;
+static int rtp_vorbis_frame_count = 1;
+
+GST_START_TEST (rtp_vorbis)
+{
+  g_print (" %s\n", __func__);
+  rtp_pipeline_test (rtp_vorbis_frame_data, rtp_vorbis_frame_data_size,
+      rtp_vorbis_frame_count, "audio/x-vorbis", "rtpvorbispay",
+      "rtpvorbisdepay");
+}
+
+GST_END_TEST
+/*
+ * Creates the test suite.
+ *
+ * Returns: pointer to the test suite.
+ */
+    Suite * rtp_create_suite ()
+{
+  Suite *s = suite_create ("rtp_data_test");
+  TCase *tc_chain = tcase_create ("linear");
+
+  /* Set timeout to 60 seconds. */
+  tcase_set_timeout (tc_chain, 60);
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, rtp_ilbc);
+  tcase_add_test (tc_chain, rtp_gsm);
+  tcase_add_test (tc_chain, rtp_amr);
+  tcase_add_test (tc_chain, rtp_pcma);
+  tcase_add_test (tc_chain, rtp_pcmu);
+  tcase_add_test (tc_chain, rtp_mpa);
+  tcase_add_test (tc_chain, rtp_h263);
+  tcase_add_test (tc_chain, rtp_h263p);
+  tcase_add_test (tc_chain, rtp_h264);
+  tcase_add_test (tc_chain, rtp_L16);
+  tcase_add_test (tc_chain, rtp_mp2t);
+  tcase_add_test (tc_chain, rtp_mp4v);
+  tcase_add_test (tc_chain, rtp_mp4g);
+  tcase_add_test (tc_chain, rtp_theora);
+  tcase_add_test (tc_chain, rtp_vorbis);
+  return s;
+}
+
+/*
+ * Main function.
+ */
+int
+main (int argc, char *argv[])
+{
+  int nf;
+
+  Suite *s = rtp_create_suite ();
+  SRunner *sr = srunner_create (s);
+
+  gst_check_init (&argc, &argv);
+
+  srunner_run_all (sr, CK_NORMAL);
+  nf = srunner_ntests_failed (sr);
+  srunner_free (sr);
+
+  return nf;
+}