Merge branch 'master' into 0.11-fdo

configure.ac

@@ -135,18 +135,20 @@ dnl *** checks for programs ***
 dnl find a compiler
 AC_PROG_CC
 AC_PROG_CC_STDC
-AC_PROG_CXX
-
-dnl determine if c++ is available on this system
-AC_CHECK_PROG(HAVE_CXX, $CXX, yes, no)
-
-dnl determine c++ preprocessor
-dnl FIXME: do we need this ?
-AC_PROG_CXXCPP
 
 dnl check if the compiler supports '-c' and '-o' options
 AM_PROG_CC_C_O
 
+dnl determine if c++ is available on this system
+AC_PROG_CXX
+dnl CXX may be set to some default even if no c++ compiler is available
+dnl (thanks autotools!), so just try to compile some c++ code to make sure
+AC_LANG_PUSH([C++])
+AC_TRY_COMPILE([ class Foo { int bar; };], , working_cxx=yes, working_cxx=no)
+AC_LANG_POP([C++])
+AC_MSG_NOTICE([working c++ compiler found: $working_cxx])
+AM_CONDITIONAL(HAVE_CXX, test "x$working_cxx" = "xyes")
+
 AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
 AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
 

ext/ogg/gstoggmux.c

@@ -835,6 +835,7 @@ gst_ogg_mux_queue_pads (GstOggMux * ogg_mux)
             GST_DEBUG_OBJECT (ogg_mux,
                 "got header buffer in control state, ignoring");
             /* just ignore */
+            pad->map.n_header_packets_seen++;
             gst_buffer_unref (buf);
             buf = NULL;
           } else {

ext/pango/gsttextoverlay.c

@@ -32,7 +32,7 @@
  * the text set via the "text" property will be rendered. If the text sink
  * pad is linked, text will be rendered as it is received on that pad,
  * honouring and matching the buffer timestamps of both input streams.
- * 
+ *
  * The text can contain newline characters and text wrapping is enabled by
  * default.
  *
@@ -41,7 +41,7 @@
  * |[
  * gst-launch -v videotestsrc ! textoverlay text="Room A" valign=top halign=left ! xvimagesink
  * ]| Here is a simple pipeline that displays a static text in the top left
- * corner of the video picture 
+ * corner of the video picture
  * |[
  * gst-launch -v filesrc location=subtitles.srt ! subparse ! txt.   videotestsrc ! timeoverlay ! textoverlay name=txt shaded-background=yes ! xvimagesink
  * ]| Here is another pipeline that displays subtitles from an .srt subtitle
@@ -54,12 +54,12 @@
  * 1
  * 00:00:03,000 --> 00:00:05,000
  * Hello? (3-5s)
- *  
+ *
  * 2
  * 00:00:08,000 --> 00:00:13,000
  * Yes, this is a subtitle. Don't
  * you like it? (8-13s)
- *  
+ *
  * 3
  * 00:00:18,826 --> 00:01:02,886
  * Uh? What are you talking about?
@@ -375,6 +375,8 @@ gst_text_overlay_class_init (GstTextOverlayClass * klass)
   gstelement_class->change_state =
       GST_DEBUG_FUNCPTR (gst_text_overlay_change_state);
 
+  klass->pango_lock = g_mutex_new ();
+
   klass->get_text = gst_text_overlay_get_text;
   fontmap = pango_cairo_font_map_get_default ();
   klass->pango_context =
@@ -1235,6 +1237,8 @@ gst_text_overlay_render_pangocairo (GstTextOverlay * overlay,
   double scalef = 1.0;
   double a, r, g, b;
 
+  g_mutex_lock (GST_TEXT_OVERLAY_GET_CLASS (overlay)->pango_lock);
+
   if (overlay->auto_adjust_size) {
     /* 640 pixel is default */
     scalef = (double) (overlay->width) / DEFAULT_SCALE_BASIS;
@@ -1250,8 +1254,8 @@ gst_text_overlay_render_pangocairo (GstTextOverlay * overlay,
 
   if (width + overlay->deltax >
       (overlay->use_vertical_render ? overlay->height : overlay->width)) {
-    /* 
+    /*
-     * subtitle image width is larger then overlay width 
+     * subtitle image width is larger then overlay width
      * so rearrange overlay wrap mode.
      */
     gst_text_overlay_update_wrap_mode (overlay);
@@ -1298,6 +1302,8 @@ gst_text_overlay_render_pangocairo (GstTextOverlay * overlay,
     cairo_matrix_init_scale (&cairo_matrix, scalef, scalef);
   }
 
+  g_mutex_unlock (GST_TEXT_OVERLAY_GET_CLASS (overlay)->pango_lock);
+
   /* reallocate surface */
   overlay->text_image = g_realloc (overlay->text_image, 4 * width * height);
 

ext/pango/gsttextoverlay.h

@@ -155,6 +155,7 @@ struct _GstTextOverlayClass {
     GstElementClass parent_class;
 
     PangoContext *pango_context;
+    GMutex       *pango_lock;
 
     gchar *     (*get_text) (GstTextOverlay *overlay, GstBuffer *video_frame);
 };

gst-libs/gst/pbutils/gstdiscoverer.c

@@ -443,6 +443,15 @@ uridecodebin_pad_added_cb (GstElement * uridecodebin, GstPad * pad,
 {
   PrivateStream *ps;
   GstPad *sinkpad = NULL;
+  GstCaps *caps;
+  static GstCaps *subs_caps = NULL;
+
+  if (!subs_caps) {
+    subs_caps = gst_caps_from_string ("text/plain; text/x-pango-markup; "
+        "subpicture/x-pgs; subpicture/x-dvb; application/x-subtitle-unknown; "
+        "application/x-ssa; application/x-ass; subtitle/x-kate; "
+        "video/x-dvd-subpicture; ");
+  }
 
   GST_DEBUG_OBJECT (dc, "pad %s:%s", GST_DEBUG_PAD_NAME (pad));
 
@@ -459,6 +468,16 @@ uridecodebin_pad_added_cb (GstElement * uridecodebin, GstPad * pad,
   g_object_set (ps->sink, "silent", TRUE, NULL);
   g_object_set (ps->queue, "max-size-buffers", 1, "silent", TRUE, NULL);
 
+  caps = gst_pad_get_caps_reffed (pad);
+
+  if (gst_caps_can_intersect (caps, subs_caps)) {
+    /* Subtitle streams are sparse and don't provide any information - don't
+     * wait for data to preroll */
+    g_object_set (ps->sink, "async", FALSE, NULL);
+  }
+
+  gst_caps_unref (caps);
+
   gst_bin_add_many (dc->priv->pipeline, ps->queue, ps->sink, NULL);
 
   if (!gst_element_link_pads_full (ps->queue, "src", ps->sink, "sink",

gst/encoding/gstencodebin.c

@@ -1698,9 +1698,9 @@ gst_encode_bin_set_profile (GstEncodeBin * ebin, GstEncodingProfile * profile)
   }
 
   /* If we're not active, we can deactivate the previous profile */
-  if (ebin->profile)
+  if (ebin->profile) {
-    gst_encoding_profile_unref (ebin->profile);
+    gst_encode_bin_tear_down_profile (ebin);
-  ebin->profile = NULL;
+  }
 
   return gst_encode_bin_setup_profile (ebin, profile);
 }

gst/playback/gstdecodebin2.c

@@ -678,7 +678,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
    * If this function returns NULL, @pad will be exposed as a final caps.
    *
    * If this function returns an empty array, the pad will be considered as
-   * having an unhandled type media type. 
+   * having an unhandled type media type.
    *
    * <note>
    *   Only the signal handler that is connected first will ever by invoked.
@@ -2696,10 +2696,12 @@ gst_decode_group_new (GstDecodeBin * dbin, GstDecodeChain * parent)
   if (G_UNLIKELY (!group->multiqueue))
     goto missing_multiqueue;
 
-  g_object_set (mq, "use-buffering", dbin->use_buffering, NULL);
+  /* default is for use-buffering is FALSE */
   if (dbin->use_buffering) {
-    g_object_set (mq, "low-percent", dbin->low_percent, NULL);
+    g_object_set (mq,
-    g_object_set (mq, "high-percent", dbin->high_percent, NULL);
+        "use-buffering", TRUE,
+        "low-percent", dbin->low_percent,
+        "high-percent", dbin->high_percent, NULL);
   }
 
   /* configure queue sizes for preroll */

gst/playback/gstplaybasebin.c

@@ -705,7 +705,7 @@ queue_threshold_reached (GstElement * queue, GstPlayBaseBin * play_base_bin)
 }
 
 /* this signal will be fired when one of the queues with raw
- * data is filled. This means that the group building stage is over 
+ * data is filled. This means that the group building stage is over
  * and playback of the new queued group should start. This is a rather unusual
  * situation because normally the group is commited when the "no_more_pads"
  * signal is fired.
@@ -843,7 +843,7 @@ gen_preroll_element (GstPlayBaseBin * play_base_bin,
 
   /* the overrun signal is always attached and serves two purposes:
    *
-   *  1) when we are building a group and the overrun is called, we commit the 
+   *  1) when we are building a group and the overrun is called, we commit the
    *     group. The reason being that if we fill the entire queue without a
    *     normal group commit (with _no_more_pads()) we can assume the
    *     audio/video is completely wacked or the element just does not know when
@@ -1252,7 +1252,7 @@ probe_triggered (GstPad * pad, GstEvent * event, gpointer user_data)
 
 /* This function will be called when the sinkpad of the preroll element
  * is unlinked, we have to connect something to the sinkpad or else the
- * state change will fail.. 
+ * state change will fail..
  */
 static void
 preroll_unlinked (GstPad * pad, GstPad * peerpad,
@@ -1315,8 +1315,8 @@ silence_stream (GstPad * pad, GstMiniObject * data, gpointer user_data)
   return FALSE;
 }
 
-/* Called by the signal handlers when a decodebin (main or subtitle) has 
+/* Called by the signal handlers when a decodebin (main or subtitle) has
- * found a new raw pad.  We create a preroll element if needed and the 
+ * found a new raw pad.  We create a preroll element if needed and the
  * appropriate streaminfo. Commits the group if there will be no more pads
  * from decodebin */
 static void
@@ -1936,7 +1936,7 @@ analyse_source (GstPlayBaseBin * play_base_bin, gboolean * is_raw,
       if (GST_PAD_TEMPLATE_DIRECTION (templ) == GST_PAD_SRC) {
         if (GST_PAD_TEMPLATE_PRESENCE (templ) == GST_PAD_SOMETIMES) {
           *is_dynamic = TRUE;
-          break;                /* only break out if we found a sometimes src pad 
+          break;                /* only break out if we found a sometimes src pad
                                    continue walking through if say a request src pad is found
                                    elements such as mpegtsparse and dvbbasebin have request
                                    and sometimes src pads */

gst/playback/gstplaybin2.c

@@ -2426,7 +2426,7 @@ notify_tags_cb (GObject * object, GParamSpec * pspec, gpointer user_data)
 }
 
 /* this function is called when a new pad is added to decodebin. We check the
- * type of the pad and add it to the selector element of the group. 
+ * type of the pad and add it to the selector element of the group.
  */
 static void
 pad_added_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
@@ -3431,7 +3431,7 @@ deactivate_group (GstPlayBin * playbin, GstSourceGroup * group)
 }
 
 /* setup the next group to play, this assumes the next_group is valid and
- * configured. It swaps out the current_group and activates the valid 
+ * configured. It swaps out the current_group and activates the valid
  * next_group. */
 static gboolean
 setup_next_source (GstPlayBin * playbin, GstState target)

gst/typefind/gsttypefindfunctions.c

@@ -1333,7 +1333,7 @@ static GstStaticCaps dts_caps = GST_STATIC_CAPS ("audio/x-dts");
 
 static gboolean
 dts_parse_frame_header (DataScanCtx * c, guint * frame_size,
-    guint * sample_rate, guint * channels)
+    guint * sample_rate, guint * channels, guint * depth, guint * endianness)
 {
   static const int sample_rates[16] = { 0, 8000, 16000, 32000, 0, 0, 11025,
     22050, 44100, 0, 0, 12000, 24000, 48000, 96000, 192000
@@ -1349,11 +1349,13 @@ dts_parse_frame_header (DataScanCtx * c, guint * frame_size,
 
   /* raw big endian or 14-bit big endian */
   if (marker == 0x7FFE8001 || marker == 0x1FFFE800) {
+    *endianness = G_BIG_ENDIAN;
     for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
       hdr[i] = GST_READ_UINT16_BE (c->data + (i * sizeof (guint16)));
   } else
     /* raw little endian or 14-bit little endian */
   if (marker == 0xFE7F0180 || marker == 0xFF1F00E8) {
+    *endianness = G_LITTLE_ENDIAN;
     for (i = 0; i < G_N_ELEMENTS (hdr); ++i)
       hdr[i] = GST_READ_UINT16_LE (c->data + (i * sizeof (guint16)));
   } else {
@@ -1376,6 +1378,9 @@ dts_parse_frame_header (DataScanCtx * c, guint * frame_size,
     hdr[5] = (hdr[5] << 12) | ((hdr[6] >> 2) & 0x0FFF);
     hdr[6] = (hdr[6] << 14) | ((hdr[7] >> 0) & 0x3FFF);
     g_assert (hdr[0] == 0x7FFE && hdr[1] == 0x8001);
+    *depth = 14;
+  } else {
+    *depth = 16;
   }
 
   GST_LOG ("frame header: %04x%04x%04x%04x", hdr[2], hdr[3], hdr[4], hdr[5]);
@@ -1409,12 +1414,13 @@ dts_type_find (GstTypeFind * tf, gpointer unused)
    * a lower probability if not found right at the start. Check that the
    * frame is followed by a second frame at the expected offset. */
   while (c.offset <= DTS_MAX_FRAMESIZE) {
-    guint frame_size = 0, rate = 0, chans = 0;
+    guint frame_size = 0, rate = 0, chans = 0, depth = 0, endianness = 0;
 
     if (G_UNLIKELY (!data_scan_ctx_ensure_data (tf, &c, DTS_MIN_FRAMESIZE)))
       return;
 
-    if (G_UNLIKELY (dts_parse_frame_header (&c, &frame_size, &rate, &chans))) {
+    if (G_UNLIKELY (dts_parse_frame_header (&c, &frame_size, &rate, &chans,
+                &depth, &endianness))) {
       GstTypeFindProbability prob;
       DataScanCtx next_c;
 
@@ -1433,10 +1439,13 @@ dts_type_find (GstTypeFind * tf, gpointer unused)
       if (chans > 0) {
         gst_type_find_suggest_simple (tf, prob, "audio/x-dts",
             "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans,
+            "depth", G_TYPE_INT, depth, "endianness", G_TYPE_INT, endianness,
             "framed", G_TYPE_BOOLEAN, FALSE, NULL);
       } else {
         gst_type_find_suggest_simple (tf, prob, "audio/x-dts",
-            "rate", G_TYPE_INT, rate, "framed", G_TYPE_BOOLEAN, FALSE, NULL);
+            "rate", G_TYPE_INT, rate, "depth", G_TYPE_INT, depth,
+            "endianness", G_TYPE_INT, endianness,
+            "framed", G_TYPE_BOOLEAN, FALSE, NULL);
       }
 
       return;

tests/check/Makefile.am

@@ -99,6 +99,12 @@ else
 check_orc =
 endif
 
+if HAVE_CXX
+cxx_checks = libs/gstlibscpp
+else
+cxx_checks =
+endif
+
 check_PROGRAMS = \
 	$(check_alsa) \
 	$(check_gnomevfs) \
@@ -144,6 +150,7 @@ check_PROGRAMS = \
 	libs/rtsp \
 	libs/tag \
 	libs/video \
+	$(cxx_checks) \
 	$(check_orc) \
 	pipelines/simple-launch-lines \
 	pipelines/streamheader \
@@ -170,7 +177,12 @@ noinst_HEADERS = \
 #	libs/struct_sparc.h \
 #	libs/struct_x86_64.h
 
-AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS) \
+AM_CFLAGS = -I$(top_srcdir)/gst-libs -I$(top_builddir)/gst-libs \
+	$(GST_CFLAGS) $(GST_CHECK_CFLAGS) \
+	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+AM_CXXFLAGS = -I$(top_srcdir)/gst-libs -I$(top_builddir)/gst-libs \
+	$(GST_CXXFLAGS) $(GST_CHECK_CFLAGS) \
 	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
 	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
 LDADD = $(GST_LIBS) $(GST_CHECK_LIBS)
@@ -295,6 +307,8 @@ libs_profile_CFLAGS = \
 libs_profile_LDADD = \
 	$(top_builddir)/gst-libs/gst/pbutils/libgstpbutils-@GST_MAJORMINOR@.la $(LDADD)
 
+libs_gstlibscpp_SOURCES = libs/gstlibscpp.cc
+
 elements_appsink_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
 	$(AM_CFLAGS)

tests/check/elements/encodebin.c

@@ -810,6 +810,66 @@ GST_START_TEST (test_encodebin_impossible_element_combination)
 
 GST_END_TEST;
 
+static void
+_test_encodebin_reuse (GstEncodingProfile * prof1, GstEncodingProfile * prof2)
+{
+  GstElement *ebin;
+
+  ebin = gst_element_factory_make ("encodebin", NULL);
+
+  /* Set a profile on encodebin... */
+  if (prof1)
+    g_object_set (ebin, "profile", prof1, NULL);
+
+  /* Make sure we can go to PAUSED */
+  fail_unless_equals_int (gst_element_set_state (ebin, GST_STATE_PAUSED),
+      GST_STATE_CHANGE_SUCCESS);
+
+  /* Set back to NULL */
+  fail_unless_equals_int (gst_element_set_state (ebin, GST_STATE_NULL),
+      GST_STATE_CHANGE_SUCCESS);
+
+  if (prof2)
+    g_object_set (ebin, "profile", prof2, NULL);
+
+  /* Make sure we can go to PLAYING */
+  fail_unless_equals_int (gst_element_set_state (ebin, GST_STATE_PAUSED),
+      GST_STATE_CHANGE_SUCCESS);
+
+  /* Set back to NULL */
+  fail_unless_equals_int (gst_element_set_state (ebin, GST_STATE_NULL),
+      GST_STATE_CHANGE_SUCCESS);
+
+  gst_object_unref (ebin);
+}
+
+GST_START_TEST (test_encodebin_reuse)
+{
+  GstEncodingProfile *prof1;
+  GstEncodingProfile *prof2;
+  GstEncodingProfile *prof3;
+  GstCaps *caps;
+
+  caps = gst_caps_new_simple ("application/ogg", NULL);
+  prof1 = (GstEncodingProfile *) gst_encoding_container_profile_new ((gchar *)
+      "myprofile", NULL, caps, NULL);
+  gst_caps_unref (caps);
+
+  prof2 = create_ogg_theora_vorbis_profile (1, 1);
+  prof3 = create_vorbis_only_profile ();
+
+  _test_encodebin_reuse (prof1, NULL);
+  _test_encodebin_reuse (prof1, prof1);
+
+  _test_encodebin_reuse (prof1, prof2);
+
+  _test_encodebin_reuse (prof2, prof3);
+
+  gst_encoding_profile_unref (prof1);
+};
+
+GST_END_TEST;
+
 
 static Suite *
 encodebin_suite (void)
@@ -831,6 +891,7 @@ encodebin_suite (void)
   tcase_add_test (tc_chain, test_encodebin_render_audio_video_static);
   tcase_add_test (tc_chain, test_encodebin_render_audio_video_dynamic);
   tcase_add_test (tc_chain, test_encodebin_impossible_element_combination);
+  tcase_add_test (tc_chain, test_encodebin_reuse);
 
   return s;
 }

tests/check/libs/.gitignore

@@ -2,6 +2,7 @@
 audio
 cddabasesrc
 fft
+gstlibscpp
 mixer
 navigation
 netbuffer

tests/check/libs/gstlibscpp.cc

@@ -0,0 +1,139 @@
+/* GStreamer
+ * Copyright (C) 2011 Tim-Philipp Müller <tim centricular net>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <gst/check/gstcheck.h>
+
+#include <gst/app/gstappbuffer.h>
+#include <gst/app/gstapp-marshal.h>
+#include <gst/app/gstappsink.h>
+#include <gst/app/gstappsrc.h>
+
+#include <gst/audio/audio-enumtypes.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudioclock.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/audio/gstaudiosink.h>
+#include <gst/audio/gstaudiosrc.h>
+#include <gst/audio/gstbaseaudiosink.h>
+#include <gst/audio/gstbaseaudiosrc.h>
+#include <gst/audio/gstringbuffer.h>
+#include <gst/audio/mixerutils.h>
+#include <gst/audio/multichannel.h>
+
+#include <gst/cdda/gstcddabasesrc.h>
+
+#include <gst/fft/gstfftf32.h>
+#include <gst/fft/gstfftf64.h>
+#include <gst/fft/gstfft.h>
+#include <gst/fft/gstffts16.h>
+#include <gst/fft/gstffts32.h>
+
+#include <gst/floatcast/floatcast.h>
+
+#include <gst/interfaces/colorbalancechannel.h>
+#include <gst/interfaces/colorbalance.h>
+#include <gst/interfaces/interfaces-enumtypes.h>
+#include <gst/interfaces/interfaces-marshal.h>
+#include <gst/interfaces/mixer.h>
+#include <gst/interfaces/mixeroptions.h>
+#include <gst/interfaces/mixertrack.h>
+#include <gst/interfaces/navigation.h>
+#include <gst/interfaces/propertyprobe.h>
+#include <gst/interfaces/streamvolume.h>
+#include <gst/interfaces/tunerchannel.h>
+#include <gst/interfaces/tuner.h>
+#include <gst/interfaces/tunernorm.h>
+#include <gst/interfaces/videoorientation.h>
+#include <gst/interfaces/xoverlay.h>
+
+#include <gst/netbuffer/gstnetbuffer.h>
+
+#include <gst/pbutils/codec-utils.h>
+#include <gst/pbutils/descriptions.h>
+#include <gst/pbutils/encoding-profile.h>
+#include <gst/pbutils/encoding-target.h>
+#include <gst/pbutils/gstdiscoverer.h>
+#include <gst/pbutils/gstpluginsbaseversion.h>
+#include <gst/pbutils/install-plugins.h>
+#include <gst/pbutils/missing-plugins.h>
+#include <gst/pbutils/pbutils-enumtypes.h>
+#include <gst/pbutils/pbutils.h>
+#include <gst/pbutils/pbutils-marshal.h>
+
+#include <gst/riff/riff-ids.h>
+#include <gst/riff/riff-media.h>
+#include <gst/riff/riff-read.h>
+
+#include <gst/rtp/gstbasertpaudiopayload.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+#include <gst/rtp/gstbasertppayload.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#include <gst/rtsp/gstrtspbase64.h>
+#include <gst/rtsp/gstrtspconnection.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtsp-enumtypes.h>
+#include <gst/rtsp/gstrtspextension.h>
+#include <gst/rtsp/gstrtsp.h>
+#include <gst/rtsp/gstrtsp-marshal.h>
+#include <gst/rtsp/gstrtspmessage.h>
+#include <gst/rtsp/gstrtsprange.h>
+#include <gst/rtsp/gstrtsptransport.h>
+#include <gst/rtsp/gstrtspurl.h>
+
+#include <gst/sdp/gstsdp.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/tag/gsttagdemux.h>
+
+#include <gst/tag/tag.h>
+
+#include <gst/video/gstvideofilter.h>
+#include <gst/video/gstvideosink.h>
+#include <gst/video/video-enumtypes.h>
+#include <gst/video/video.h>
+
+/* we mostly just want to make sure that our library headers don't
+ * contain anything a C++ compiler might not like */
+GST_START_TEST (test_nothing)
+{
+  gst_init (NULL, NULL);
+}
+
+GST_END_TEST;
+
+static Suite *
+libscpp_suite (void)
+{
+  Suite *s = suite_create ("GstLibsCpp");
+  TCase *tc_chain = tcase_create ("C++ libs header tests");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_nothing);
+
+  return s;
+}
+
+GST_CHECK_MAIN (libscpp);