Merge branch 'master' into 0.11

Conflicts: ext/cairo/gsttextoverlay.c ext/pulse/pulseaudiosink.c gst/audioparsers/gstaacparse.c gst/avi/gstavimux.c gst/flv/gstflvmux.c gst/interleave/interleave.c gst/isomp4/gstqtmux.c gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/matroska/matroska-mux.h gst/matroska/matroska-read-common.c gst/multifile/gstmultifilesink.c gst/multipart/multipartmux.c gst/shapewipe/gstshapewipe.c gst/smpte/gstsmpte.c gst/udp/gstmultiudpsink.c gst/videobox/gstvideobox.c gst/videocrop/gstaspectratiocrop.c gst/videomixer/videomixer.c gst/videomixer/videomixer2.c gst/wavparse/gstwavparse.c po/ja.po po/lv.po po/sr.po tests/check/Makefile.am tests/check/elements/qtmux.c tests/check/elements/rgvolume.c
author: Sebastian Dröge <sebastian.droege@collabora.co.uk> 2012-01-10 14:32:32 +0100
committer: Sebastian Dröge <sebastian.droege@collabora.co.uk> 2012-01-10 14:32:32 +0100
commit: 93e3ed5a869afb79f9658f22ced4691ce475e8a1 (patch)
tree: baaec96daaf22857d61128401c12e2649ef25304
parent: 692e00cc00d772fe2b01f518cb9ac88d124807d1 (diff)
parent: 2b2c0940f1b7ce8a858a26ad5e246fd645a83830 (diff)
85 files changed, 2635 insertions, 681 deletions
diff --git a/Makefile.am b/Makefile.am
index 0119fad4c..9638883cb 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -26,7 +26,8 @@ DISTCLEANFILES = _stdint.h
 
 noinst_HEADERS = \
 	gst-libs/gst/gettext.h \
-	gst-libs/gst/gst-i18n-plugin.h
+	gst-libs/gst/gst-i18n-plugin.h \
+	gst-libs/gst/glib-compat-private.h
 
 ACLOCAL_AMFLAGS = -I m4 -I common/m4
 
diff --git a/configure.ac b/configure.ac
index f2e608bda..5e16f67b1 100644
--- a/configure.ac
+++ b/configure.ac
@@ -203,7 +203,7 @@ AC_CHECK_TYPE([struct ip_mreqn], [
 dnl *** checks for dependency libraries ***
 
 dnl GLib is required
-AG_GST_GLIB_CHECK([2.20])
+AG_GST_GLIB_CHECK([2.24])
 PKG_CHECK_MODULES(GIO, [ gio-2.0 >= 2.20 ], , AC_MSG_ERROR([gio is required]))
 
 dnl Orc
@@ -438,6 +438,12 @@ int main ()
   AC_SUBST(HAVE_DIRECTSOUND)
 ])
 
+dnl *** Win32 WaveOut ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_WAVEFORM, true)
+AG_GST_CHECK_FEATURE(WAVEFORM, [Win32 WaveForm], waveformsink, [
+  AC_CHECK_HEADER(mmsystem.h, HAVE_WAVEFORM="yes", HAVE_WAVEFORM="no", [#include <windows.h>])
+])
+
 dnl *** OSS audio *** (Linux, *BSD)
 translit(dnm, m, l) AM_CONDITIONAL(USE_OSS, true)
 AG_GST_CHECK_FEATURE(OSS, [OSS audio], ossaudio, [
@@ -1005,6 +1011,7 @@ AM_CONDITIONAL(USE_SOUP, false)
 AM_CONDITIONAL(USE_SPEEX, false)
 AM_CONDITIONAL(USE_SUNAUDIO, false)
 AM_CONDITIONAL(USE_TAGLIB, false)
+AM_CONDITIONAL(USE_WAVEFORM, false)
 AM_CONDITIONAL(USE_WAVPACK, false)
 AM_CONDITIONAL(USE_X, false)
 AM_CONDITIONAL(USE_XSHM, false)
diff --git a/ext/cairo/gsttextoverlay.c b/ext/cairo/gsttextoverlay.c
index 7d79f98f7..2a09b9cb1 100644
--- a/ext/cairo/gsttextoverlay.c
+++ b/ext/cairo/gsttextoverlay.c
@@ -106,7 +106,7 @@ static gboolean gst_text_overlay_setcaps (GstPad * pad, GstCaps * caps);
 static GstPadLinkReturn gst_text_overlay_text_pad_linked (GstPad * pad,
     GstPad * peer);
 static void gst_text_overlay_text_pad_unlinked (GstPad * pad);
-static GstFlowReturn gst_text_overlay_collected (GstCollectPads * pads,
+static GstFlowReturn gst_text_overlay_collected (GstCollectPads2 * pads,
     gpointer data);
 static void gst_text_overlay_finalize (GObject * object);
 static void gst_text_overlay_font_init (GstCairoTextOverlay * overlay);
@@ -215,7 +215,7 @@ gst_text_overlay_finalize (GObject * object)
 {
   GstCairoTextOverlay *overlay = GST_CAIRO_TEXT_OVERLAY (object);
 
-  gst_collect_pads_stop (overlay->collect);
+  gst_collect_pads2_stop (overlay->collect);
   gst_object_unref (overlay->collect);
 
   g_free (overlay->text_fill_image);
@@ -279,16 +279,16 @@ gst_text_overlay_init (GstCairoTextOverlay * overlay,
   overlay->fps_n = 0;
   overlay->fps_d = 1;
 
-  overlay->collect = gst_collect_pads_new ();
+  overlay->collect = gst_collect_pads2_new ();
 
-  gst_collect_pads_set_function (overlay->collect,
+  gst_collect_pads2_set_function (overlay->collect,
       GST_DEBUG_FUNCPTR (gst_text_overlay_collected), overlay);
 
-  overlay->video_collect_data = gst_collect_pads_add_pad (overlay->collect,
-      overlay->video_sinkpad, sizeof (GstCollectData), NULL);
+  overlay->video_collect_data = gst_collect_pads2_add_pad (overlay->collect,
+      overlay->video_sinkpad, sizeof (GstCollectData2));
 
   /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
+   * GstCollectPads2; because it sets its own event function giving the
    * element no access to events. Nicked from avimux. */
   overlay->collect_event =
       (GstPadEventFunction) GST_PAD_EVENTFUNC (overlay->video_sinkpad);
@@ -640,8 +640,8 @@ gst_text_overlay_text_pad_linked (GstPad * pad, GstPad * peer)
   GST_DEBUG_OBJECT (overlay, "Text pad linked");
 
   if (overlay->text_collect_data == NULL) {
-    overlay->text_collect_data = gst_collect_pads_add_pad (overlay->collect,
-        overlay->text_sinkpad, sizeof (GstCollectData), NULL);
+    overlay->text_collect_data = gst_collect_pads2_add_pad (overlay->collect,
+        overlay->text_sinkpad, sizeof (GstCollectData2));
   }
 
   overlay->need_render = TRUE;
@@ -660,7 +660,7 @@ gst_text_overlay_text_pad_unlinked (GstPad * pad)
   GST_DEBUG_OBJECT (overlay, "Text pad unlinked");
 
   if (overlay->text_collect_data) {
-    gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad);
+    gst_collect_pads2_remove_pad (overlay->collect, overlay->text_sinkpad);
     overlay->text_collect_data = NULL;
   }
 
@@ -807,7 +807,7 @@ gst_text_overlay_pop_video (GstCairoTextOverlay * overlay)
 {
   GstBuffer *buf;
 
-  buf = gst_collect_pads_pop (overlay->collect, overlay->video_collect_data);
+  buf = gst_collect_pads2_pop (overlay->collect, overlay->video_collect_data);
   g_return_if_fail (buf != NULL);
   gst_buffer_unref (buf);
 }
@@ -818,7 +818,7 @@ gst_text_overlay_pop_text (GstCairoTextOverlay * overlay)
   GstBuffer *buf;
 
   if (overlay->text_collect_data) {
-    buf = gst_collect_pads_pop (overlay->collect, overlay->text_collect_data);
+    buf = gst_collect_pads2_pop (overlay->collect, overlay->text_collect_data);
     g_return_if_fail (buf != NULL);
     gst_buffer_unref (buf);
   }
@@ -828,7 +828,7 @@ gst_text_overlay_pop_text (GstCairoTextOverlay * overlay)
 
 /* This function is called when there is data on all pads */
 static GstFlowReturn
-gst_text_overlay_collected (GstCollectPads * pads, gpointer data)
+gst_text_overlay_collected (GstCollectPads2 * pads, gpointer data)
 {
   GstCairoTextOverlay *overlay;
   GstFlowReturn ret = GST_FLOW_OK;
@@ -842,14 +842,14 @@ gst_text_overlay_collected (GstCollectPads * pads, gpointer data)
 
   GST_DEBUG ("Collecting");
 
-  video_frame = gst_collect_pads_peek (overlay->collect,
+  video_frame = gst_collect_pads2_peek (overlay->collect,
       overlay->video_collect_data);
 
   /* send EOS if video stream EOSed regardless of text stream */
   if (video_frame == NULL) {
     GST_DEBUG ("Video stream at EOS");
     if (overlay->text_collect_data) {
-      text_buf = gst_collect_pads_pop (overlay->collect,
+      text_buf = gst_collect_pads2_pop (overlay->collect,
           overlay->text_collect_data);
     }
     gst_pad_push_event (overlay->srcpad, gst_event_new_eos ());
@@ -892,7 +892,7 @@ gst_text_overlay_collected (GstCollectPads * pads, gpointer data)
     goto done;
   }
 
-  text_buf = gst_collect_pads_peek (overlay->collect,
+  text_buf = gst_collect_pads2_peek (overlay->collect,
       overlay->text_collect_data);
 
   /* just push the video frame if the text stream has EOSed */
@@ -1004,7 +1004,7 @@ gst_text_overlay_video_event (GstPad * pad, GstEvent * event)
     gst_pad_push_event (overlay->srcpad, event);
   }
 
-  /* now GstCollectPads can take care of the rest, e.g. EOS */
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
   ret = overlay->collect_event (pad, event);
   gst_object_unref (overlay);
   return ret;
@@ -1018,12 +1018,12 @@ gst_text_overlay_change_state (GstElement * element, GstStateChange transition)
 
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_collect_pads_start (overlay->collect);
+      gst_collect_pads2_start (overlay->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       /* need to unblock the collectpads before calling the
        * parent change_state so that streaming can finish */
-      gst_collect_pads_stop (overlay->collect);
+      gst_collect_pads2_stop (overlay->collect);
       break;
     default:
       break;
diff --git a/ext/cairo/gsttextoverlay.h b/ext/cairo/gsttextoverlay.h
index dbb21546f..1f98d3d58 100644
--- a/ext/cairo/gsttextoverlay.h
+++ b/ext/cairo/gsttextoverlay.h
@@ -3,7 +3,7 @@
 #define __GST_CAIRO_TEXT_OVERLAY_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 G_BEGIN_DECLS
 
@@ -45,9 +45,9 @@ struct _GstCairoTextOverlay {
     GstPad               *text_sinkpad;
     GstPad               *srcpad;
 
-    GstCollectPads       *collect;
-    GstCollectData       *video_collect_data;
-    GstCollectData       *text_collect_data;
+    GstCollectPads2      *collect;
+    GstCollectData2      *video_collect_data;
+    GstCollectData2      *text_collect_data;
     GstPadEventFunction   collect_event;
 
     gint                  width;
diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c
index d236c3ffa..d3618ed19 100644
--- a/ext/dv/gstdvdemux.c
+++ b/ext/dv/gstdvdemux.c
@@ -21,6 +21,11 @@
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <math.h>
 
diff --git a/ext/flac/gstflacdec.c b/ext/flac/gstflacdec.c
index 94c0402cc..7066d7db9 100644
--- a/ext/flac/gstflacdec.c
+++ b/ext/flac/gstflacdec.c
@@ -41,6 +41,11 @@
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 
 #include "gstflacdec.h"
diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c
index f18d3c183..923538677 100644
--- a/ext/gdk_pixbuf/gstgdkpixbufsink.c
+++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c
@@ -271,7 +271,7 @@ gst_gdk_pixbuf_sink_set_caps (GstBaseSink * basesink, GstCaps * caps)
 
   GST_INFO_OBJECT (sink, "format             : %d", fmt);
   GST_INFO_OBJECT (sink, "width x height     : %d x %d", w, h);
-  GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_d, par_n);
+  GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_n, par_d);
 
   return TRUE;
 }
@@ -344,7 +344,7 @@ gst_gdk_pixbuf_sink_handle_buffer (GstBaseSink * basesink, GstBuffer * buf,
      * The structure will take its own ref to the pixbuf. */
     s = gst_structure_new (msg_name,
         "pixbuf", GDK_TYPE_PIXBUF, pixbuf,
-        "pixel-aspect-ratio", GST_TYPE_FRACTION, sink->par_d, sink->par_n,
+        "pixel-aspect-ratio", GST_TYPE_FRACTION, sink->par_n, sink->par_d,
         NULL);
 
     msg = gst_message_new_element (GST_OBJECT_CAST (sink), s);
diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c
index 1789edb60..2bb355529 100644
--- a/ext/jack/gstjackaudioclient.c
+++ b/ext/jack/gstjackaudioclient.c
@@ -23,6 +23,8 @@
 
 #include "gstjackaudioclient.h"
 
+#include <gst/glib-compat-private.h>
+
 GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
 #define GST_CAT_DEFAULT gst_jack_audio_client_debug
 
diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
index fe911840f..8055f9219 100644
--- a/ext/jack/gstjackaudiosink.c
+++ b/ext/jack/gstjackaudiosink.c
@@ -329,7 +329,11 @@ gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (sink, "open");
 
-  name = g_get_application_name ();
+  if (sink->client_name) {
+    name = sink->client_name;
+  } else {
+    name = g_get_application_name ();
+  }
   if (!name)
     name = "GStreamer";
 
@@ -648,8 +652,9 @@ enum
   SIGNAL_LAST
 };
 
-#define DEFAULT_PROP_CONNECT 	GST_JACK_CONNECT_AUTO
-#define DEFAULT_PROP_SERVER 	NULL
+#define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 		NULL
+#define DEFAULT_PROP_CLIENT_NAME 	NULL
 
 enum
 {
@@ -657,6 +662,7 @@ enum
   PROP_CONNECT,
   PROP_SERVER,
   PROP_CLIENT,
+  PROP_CLIENT_NAME,
   PROP_LAST
 };
 
@@ -705,6 +711,19 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
           "The Jack server to connect to (NULL = default)",
           DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstJackAudioSink:client-name
+   *
+   * The client name to use.
+   *
+   * Since: 0.10.31
+   */
+  g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
+      g_param_spec_string ("client-name", "Client name",
+          "The client name of the Jack instance (NULL = default)",
+          DEFAULT_PROP_CLIENT_NAME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_CLIENT,
       g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
           GST_TYPE_JACK_CLIENT,
@@ -738,6 +757,7 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink)
   sink->jclient = NULL;
   sink->ports = NULL;
   sink->port_count = 0;
+  sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
   sink->buffers = NULL;
 }
 
@@ -747,6 +767,12 @@ gst_jack_audio_sink_dispose (GObject * object)
   GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
 
   gst_caps_replace (&sink->caps, NULL);
+
+  if (sink->client_name != NULL) {
+    g_free (sink->client_name);
+    sink->client_name = NULL;
+  }
+
   G_OBJECT_CLASS (parent_class)->dispose (object);
 }
 
@@ -759,6 +785,10 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
   sink = GST_JACK_AUDIO_SINK (object);
 
   switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_free (sink->client_name);
+      sink->client_name = g_value_dup_string (value);
+      break;
     case PROP_CONNECT:
       sink->connect = g_value_get_enum (value);
       break;
@@ -787,6 +817,9 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
   sink = GST_JACK_AUDIO_SINK (object);
 
   switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_value_set_string (value, sink->client_name);
+      break;
     case PROP_CONNECT:
       g_value_set_enum (value, sink->connect);
       break;
diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h
index 02a8db9fc..0dbc142e0 100644
--- a/ext/jack/gstjackaudiosink.h
+++ b/ext/jack/gstjackaudiosink.h
@@ -58,6 +58,7 @@ struct _GstJackAudioSink {
   GstJackConnect   connect;
   gchar           *server;
   jack_client_t   *jclient;
+  gchar           *client_name;
 
   /* our client */
   GstJackAudioClient *client;
diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c
index 4973a8b77..cf21cbfc1 100644
--- a/ext/jack/gstjackaudiosrc.c
+++ b/ext/jack/gstjackaudiosrc.c
@@ -336,7 +336,11 @@ gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (src, "open");
 
-  name = g_get_application_name ();
+  if (src->client_name) {
+    name = src->client_name;
+  } else {
+    name = g_get_application_name ();
+  }
   if (!name)
     name = "GStreamer";
 
@@ -651,8 +655,9 @@ enum
   LAST_SIGNAL
 };
 
-#define DEFAULT_PROP_CONNECT 	GST_JACK_CONNECT_AUTO
-#define DEFAULT_PROP_SERVER 	NULL
+#define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 		NULL
+#define DEFAULT_PROP_CLIENT_NAME	NULL
 
 enum
 {
@@ -660,6 +665,7 @@ enum
   PROP_CONNECT,
   PROP_SERVER,
   PROP_CLIENT,
+  PROP_CLIENT_NAME,
   PROP_LAST
 };
 
@@ -726,6 +732,19 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
           "The Jack server to connect to (NULL = default)",
           DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstJackAudioSrc:client-name
+   *
+   * The client name to use.
+   *
+   * Since: 0.10.31
+   */
+  g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
+      g_param_spec_string ("client-name", "Client name",
+          "The client name of the Jack instance (NULL = default)",
+          DEFAULT_PROP_CLIENT_NAME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_CLIENT,
       g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
           GST_TYPE_JACK_CLIENT,
@@ -765,6 +784,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src)
   src->ports = NULL;
   src->port_count = 0;
   src->buffers = NULL;
+  src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
 }
 
 static void
@@ -773,6 +793,12 @@ gst_jack_audio_src_dispose (GObject * object)
   GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
 
   gst_caps_replace (&src->caps, NULL);
+
+  if (src->client_name != NULL) {
+    g_free (src->client_name);
+    src->client_name = NULL;
+  }
+
   G_OBJECT_CLASS (parent_class)->dispose (object);
 }
 
@@ -783,6 +809,10 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id,
   GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
 
   switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_free (src->client_name);
+      src->client_name = g_value_dup_string (value);
+      break;
     case PROP_CONNECT:
       src->connect = g_value_get_enum (value);
       break;
@@ -809,6 +839,9 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id,
   GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
 
   switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_value_set_string (value, src->client_name);
+      break;
     case PROP_CONNECT:
       g_value_set_enum (value, src->connect);
       break;
diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h
index fa119472c..1401d9b0d 100644
--- a/ext/jack/gstjackaudiosrc.h
+++ b/ext/jack/gstjackaudiosrc.h
@@ -75,6 +75,7 @@ struct _GstJackAudioSrc
     GstJackConnect connect;
     gchar           *server;
     jack_client_t   *jclient;
+    gchar           *client_name;
 
     /* our client */
     GstJackAudioClient *client;
diff --git a/ext/jpeg/gstjpegdec.c b/ext/jpeg/gstjpegdec.c
index 15658ed25..8301214ec 100644
--- a/ext/jpeg/gstjpegdec.c
+++ b/ext/jpeg/gstjpegdec.c
@@ -761,7 +761,7 @@ gst_jpeg_dec_getcaps (GstPad * pad, GstCaps * filter)
     templ_caps = gst_pad_get_pad_template_caps (pad);
     caps = gst_caps_intersect_full (peer_caps, templ_caps,
         GST_CAPS_INTERSECT_FIRST);
-
+    gst_caps_unref (peer_caps);
     gst_object_unref (peer);
   } else {
     caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c
index 4fa873c4d..371863e7a 100644
--- a/ext/pulse/pulsesink.c
+++ b/ext/pulse/pulsesink.c
@@ -58,6 +58,8 @@
 
 #include <gst/pbutils/pbutils.h>        /* only used for GST_PLUGINS_BASE_VERSION_* */
 
+#include <gst/glib-compat-private.h>
+
 #include "pulsesink.h"
 #include "pulseutil.h"
 
diff --git a/ext/soup/gstsouphttpclientsink.c b/ext/soup/gstsouphttpclientsink.c
index af238277f..d8f46df9d 100644
--- a/ext/soup/gstsouphttpclientsink.c
+++ b/ext/soup/gstsouphttpclientsink.c
@@ -42,6 +42,8 @@
 #include <gst/base/gstbasesink.h>
 #include "gstsouphttpclientsink.h"
 
+#include <gst/glib-compat-private.h>
+
 GST_DEBUG_CATEGORY_STATIC (souphttpclientsink_dbg);
 #define GST_CAT_DEFAULT souphttpclientsink_dbg
 
@@ -445,6 +447,20 @@ gst_soup_http_client_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
 
 }
 
+static gboolean
+thread_ready_idle_cb (gpointer data)
+{
+  GstSoupHttpClientSink *souphttpsink = GST_SOUP_HTTP_CLIENT_SINK (data);
+
+  GST_LOG_OBJECT (souphttpsink, "thread ready");
+
+  g_mutex_lock (souphttpsink->mutex);
+  g_cond_signal (souphttpsink->cond);
+  g_mutex_unlock (souphttpsink->mutex);
+
+  return FALSE;                 /* only run once */
+}
+
 static gpointer
 thread_func (gpointer ptr)
 {
@@ -452,7 +468,6 @@ thread_func (gpointer ptr)
 
   GST_DEBUG ("thread start");
 
-  souphttpsink->loop = g_main_loop_new (souphttpsink->context, TRUE);
   g_main_loop_run (souphttpsink->loop);
 
   GST_DEBUG ("thread quit");
@@ -468,12 +483,35 @@ gst_soup_http_client_sink_start (GstBaseSink * sink)
   if (souphttpsink->prop_session) {
     souphttpsink->session = souphttpsink->prop_session;
   } else {
+    GSource *source;
     GError *error = NULL;
 
     souphttpsink->context = g_main_context_new ();
 
+    /* set up idle source to signal when the main loop is running and
+     * it's safe for ::stop() to call g_main_loop_quit() */
+    source = g_idle_source_new ();
+    g_source_set_callback (source, thread_ready_idle_cb, sink, NULL);
+    g_source_attach (source, souphttpsink->context);
+    g_source_unref (source);
+
+    souphttpsink->loop = g_main_loop_new (souphttpsink->context, TRUE);
+
+    g_mutex_lock (souphttpsink->mutex);
+
+    /* FIXME: error handling */
+#if !GLIB_CHECK_VERSION (2, 31, 0)
     souphttpsink->thread = g_thread_create (thread_func, souphttpsink,
         TRUE, &error);
+#else
+    souphttpsink->thread = g_thread_try_new ("souphttpclientsink-thread",
+        thread_func, souphttpsink, &error);
+#endif
+
+    GST_LOG_OBJECT (souphttpsink, "waiting for main loop thread to start up");
+    g_cond_wait (souphttpsink->cond, souphttpsink->mutex);
+    g_mutex_unlock (souphttpsink->mutex);
+    GST_LOG_OBJECT (souphttpsink, "main loop thread running");
 
     souphttpsink->session =
         soup_session_async_new_with_options (SOUP_SESSION_ASYNC_CONTEXT,
diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c
index 50e1df933..4bb1e7e2b 100644
--- a/ext/wavpack/gstwavpackparse.c
+++ b/ext/wavpack/gstwavpackparse.c
@@ -41,6 +41,11 @@
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <gst/gst.h>
 #include <gst/gst-i18n-plugin.h>
 
diff --git a/gst-libs/gst/glib-compat-private.h b/gst-libs/gst/glib-compat-private.h
new file mode 100644
index 000000000..b9248e686
--- /dev/null
+++ b/gst-libs/gst/glib-compat-private.h
@@ -0,0 +1,135 @@
+/*
+ * glib-compat.c
+ * Functions copied from glib 2.10
+ *
+ * Copyright 2005 David Schleef <ds@schleef.org>
+ *
+ * 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.
+ */
+
+#ifndef __GLIB_COMPAT_PRIVATE_H__
+#define __GLIB_COMPAT_PRIVATE_H__
+
+#include <glib.h>
+
+G_BEGIN_DECLS
+
+#if !GLIB_CHECK_VERSION(2,25,0)
+
+#if defined (_MSC_VER) && !defined(_WIN64)
+typedef struct _stat32 GStatBuf;
+#else
+typedef struct stat GStatBuf;
+#endif
+
+#endif
+
+#if GLIB_CHECK_VERSION(2,26,0)
+#define GLIB_HAS_GDATETIME
+#endif
+
+/* See bug #651514 */
+#if GLIB_CHECK_VERSION(2,29,5)
+#define G_ATOMIC_POINTER_COMPARE_AND_EXCHANGE(a,b,c) \
+    g_atomic_pointer_compare_and_exchange ((a),(b),(c))
+#define G_ATOMIC_INT_COMPARE_AND_EXCHANGE(a,b,c) \
+    g_atomic_int_compare_and_exchange ((a),(b),(c))
+#else
+#define G_ATOMIC_POINTER_COMPARE_AND_EXCHANGE(a,b,c) \
+    g_atomic_pointer_compare_and_exchange ((volatile gpointer *)(a),(b),(c))
+#define G_ATOMIC_INT_COMPARE_AND_EXCHANGE(a,b,c) \
+    g_atomic_int_compare_and_exchange ((volatile int *)(a),(b),(c))
+#endif
+
+/* See bug #651514 */
+#if GLIB_CHECK_VERSION(2,29,5)
+#define G_ATOMIC_INT_ADD(a,b) g_atomic_int_add ((a),(b))
+#else
+#define G_ATOMIC_INT_ADD(a,b) g_atomic_int_exchange_and_add ((a),(b))
+#endif
+
+/* copies */
+
+#if GLIB_CHECK_VERSION (2, 31, 0)
+#define g_mutex_new gst_g_mutex_new
+static inline GMutex *
+gst_g_mutex_new (void)
+{
+  GMutex *mutex = g_slice_new (GMutex);
+  g_mutex_init (mutex);
+  return mutex;
+}
+#define g_mutex_free gst_g_mutex_free
+static inline void
+gst_g_mutex_free (GMutex *mutex)
+{
+  g_mutex_clear (mutex);
+  g_slice_free (GMutex, mutex);
+}
+#define g_static_rec_mutex_init gst_g_static_rec_mutex_init
+static inline void
+gst_g_static_rec_mutex_init (GStaticRecMutex *mutex)
+{
+  static const GStaticRecMutex init_mutex = G_STATIC_REC_MUTEX_INIT;
+
+  *mutex = init_mutex;
+}
+#define g_cond_new gst_g_cond_new
+static inline GCond *
+gst_g_cond_new (void)
+{
+  GCond *cond = g_slice_new (GCond);
+  g_cond_init (cond);
+  return cond;
+}
+#define g_cond_free gst_g_cond_free
+static inline void
+gst_g_cond_free (GCond *cond)
+{
+  g_cond_clear (cond);
+  g_slice_free (GCond, cond);
+}
+#define g_cond_timed_wait gst_g_cond_timed_wait
+static inline gboolean
+gst_g_cond_timed_wait (GCond *cond, GMutex *mutex, GTimeVal *abs_time)
+{
+  gint64 end_time;
+
+  if (abs_time == NULL) {
+    g_cond_wait (cond, mutex);
+    return TRUE;
+  }
+
+  end_time = abs_time->tv_sec;
+  end_time *= 1000000;
+  end_time += abs_time->tv_usec;
+
+  /* would be nice if we had clock_rtoffset, but that didn't seem to
+   * make it into the kernel yet...
+   */
+  /* if CLOCK_MONOTONIC is not defined then g_get_montonic_time() and
+   * g_get_real_time() are returning the same clock and we'd add ~0
+   */
+  end_time += g_get_monotonic_time () - g_get_real_time ();
+  return g_cond_wait_until (cond, mutex, end_time);
+}
+#endif /* GLIB_CHECK_VERSION (2, 31, 0) */
+
+/* adaptations */
+
+G_END_DECLS
+
+#endif
diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c
index c802c64e6..325bdc53b 100644
--- a/gst/alpha/gstalpha.c
+++ b/gst/alpha/gstalpha.c
@@ -145,6 +145,8 @@ GST_STATIC_PAD_TEMPLATE ("sink",
 static GstStaticCaps gst_alpha_alpha_caps =
 GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("{ AYUV, ARGB, BGRA, ABGR, RGBA }"));
 
+/* FIXME: why do we need our own lock for this? */
+#if !GLIB_CHECK_VERSION (2, 31, 0)
 #define GST_ALPHA_LOCK(alpha) G_STMT_START { \
   GST_LOG_OBJECT (alpha, "Locking alpha from thread %p", g_thread_self ()); \
   g_static_mutex_lock (&alpha->lock); \
@@ -155,6 +157,18 @@ GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("{ AYUV, ARGB, BGRA, ABGR, RGBA }"));
   GST_LOG_OBJECT (alpha, "Unlocking alpha from thread %p", g_thread_self ()); \
   g_static_mutex_unlock (&alpha->lock); \
 } G_STMT_END
+#else
+#define GST_ALPHA_LOCK(alpha) G_STMT_START { \
+  GST_LOG_OBJECT (alpha, "Locking alpha from thread %p", g_thread_self ()); \
+  g_mutex_lock (&alpha->lock); \
+  GST_LOG_OBJECT (alpha, "Locked alpha from thread %p", g_thread_self ()); \
+} G_STMT_END
+
+#define GST_ALPHA_UNLOCK(alpha) G_STMT_START { \
+  GST_LOG_OBJECT (alpha, "Unlocking alpha from thread %p", g_thread_self ()); \
+  g_mutex_unlock (&alpha->lock); \
+} G_STMT_END
+#endif
 
 static GstCaps *gst_alpha_transform_caps (GstBaseTransform * btrans,
     GstPadDirection direction, GstCaps * caps, GstCaps * filter);
@@ -296,7 +310,11 @@ gst_alpha_init (GstAlpha * alpha)
   alpha->black_sensitivity = DEFAULT_BLACK_SENSITIVITY;
   alpha->white_sensitivity = DEFAULT_WHITE_SENSITIVITY;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   g_static_mutex_init (&alpha->lock);
+#else
+  g_mutex_init (&alpha->lock);
+#endif
 }
 
 static void
@@ -304,7 +322,11 @@ gst_alpha_finalize (GObject * object)
 {
   GstAlpha *alpha = GST_ALPHA (object);
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   g_static_mutex_free (&alpha->lock);
+#else
+  g_mutex_clear (&alpha->lock);
+#endif
 
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
diff --git a/gst/alpha/gstalpha.h b/gst/alpha/gstalpha.h
index c24b753ba..48592f1bb 100644
--- a/gst/alpha/gstalpha.h
+++ b/gst/alpha/gstalpha.h
@@ -69,7 +69,11 @@ struct _GstAlpha
   /* <private> */
 
   /* caps */
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   GStaticMutex lock;
+#else
+  GMutex lock;
+#endif
 
   gboolean in_sdtv, out_sdtv;
 
diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c
index f0d2e34f0..cf431618e 100644
--- a/gst/audiofx/audiochebband.c
+++ b/gst/audiofx/audiochebband.c
@@ -83,6 +83,8 @@
 
 #include "audiochebband.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_audio_cheb_band_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c
index 250857ec0..9654a0564 100644
--- a/gst/audiofx/audiocheblimit.c
+++ b/gst/audiofx/audiocheblimit.c
@@ -79,6 +79,8 @@
 
 #include "audiocheblimit.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c
index 414768920..9fa21111b 100644
--- a/gst/audiofx/audiofirfilter.c
+++ b/gst/audiofx/audiofirfilter.c
@@ -56,6 +56,8 @@
 
 #include "audiofirfilter.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_audio_fir_filter_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c
index f411d8871..dc3c17efb 100644
--- a/gst/audiofx/audioiirfilter.c
+++ b/gst/audiofx/audioiirfilter.c
@@ -52,6 +52,8 @@
 
 #include "audioiirfilter.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_audio_iir_filter_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 3f23b561a..cd50b0655 100644
--- a/gst/audiofx/audiowsincband.c
+++ b/gst/audiofx/audiowsincband.c
@@ -63,6 +63,8 @@
 
 #include "audiowsincband.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c
index f13d2d020..7a9f5713b 100644
--- a/gst/audiofx/audiowsinclimit.c
+++ b/gst/audiofx/audiowsinclimit.c
@@ -63,6 +63,8 @@
 
 #include "audiowsinclimit.h"
 
+#include "gst/glib-compat-private.h"
+
 #define GST_CAT_DEFAULT gst_audio_wsinclimit_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c
index 5c0d88a06..789731599 100644
--- a/gst/audioparsers/gstaacparse.c
+++ b/gst/audioparsers/gstaacparse.c
@@ -44,6 +44,7 @@
 
 #include <string.h>
 
+#include <gst/base/gstbitreader.h>
 #include "gstaacparse.h"
 
 
@@ -52,7 +53,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/mpeg, "
         "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, "
-        "stream-format = (string) { raw, adts, adif };"));
+        "stream-format = (string) { raw, adts, adif, loas };"));
 
 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
@@ -65,10 +66,21 @@ GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
 
 #define ADIF_MAX_SIZE 40        /* Should be enough */
 #define ADTS_MAX_SIZE 10        /* Should be enough */
+#define LOAS_MAX_SIZE 3         /* Should be enough */
 
 
 #define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
 
+static const gint loas_sample_rate_table[32] = {
+  96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+  16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+static const gint loas_channels_table[32] = {
+  0, 1, 2, 3, 4, 5, 6, 8,
+  0, 0, 0, 0, 0, 0, 0, 0
+};
+
 static gboolean gst_aac_parse_start (GstBaseParse * parse);
 static gboolean gst_aac_parse_stop (GstBaseParse * parse);
 
@@ -190,6 +202,9 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
     case DSPAAC_HEADER_ADIF:
       stream_format = "adif";
       break;
+    case DSPAAC_HEADER_LOAS:
+      stream_format = "loas";
+      break;
     default:
       stream_format = NULL;
   }
@@ -329,6 +344,8 @@ gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
     const guint8 * data, const guint avail, gboolean drain,
     guint * framesize, guint * needed_data)
 {
+  *needed_data = 0;
+
   if (G_UNLIKELY (avail < 2))
     return FALSE;
 
@@ -367,6 +384,292 @@ gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
   return FALSE;
 }
 
+static gboolean
+gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
+    guint32 * value)
+{
+  guint8 bytes, i, byte;
+
+  *value = 0;
+  if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
+    return FALSE;
+  for (i = 0; i < bytes; ++i) {
+    *value <<= 8;
+    if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
+      return FALSE;
+    *value += byte;
+  }
+  return TRUE;
+}
+
+static gboolean
+gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
+    guint8 * audio_object_type)
+{
+  if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
+    return FALSE;
+  if (*audio_object_type == 31) {
+    if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
+      return FALSE;
+    *audio_object_type += 32;
+  }
+  GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
+  return TRUE;
+}
+
+static gboolean
+gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
+    gint * sample_rate)
+{
+  guint8 sampling_frequency_index;
+  if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
+    return FALSE;
+  GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
+      sampling_frequency_index);
+  if (sampling_frequency_index == 0xf) {
+    guint32 sampling_rate;
+    if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
+      return FALSE;
+    *sample_rate = sampling_rate;
+  } else {
+    *sample_rate = loas_sample_rate_table[sampling_frequency_index];
+    if (!*sample_rate)
+      return FALSE;
+  }
+  return TRUE;
+}
+
+/* See table 1.13 in ISO/IEC 14496-3 */
+static gboolean
+gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
+    GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
+{
+  guint8 audio_object_type, channel_configuration;
+
+  if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
+    return FALSE;
+
+  if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
+    return FALSE;
+
+  if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
+    return FALSE;
+  GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
+  *channels = loas_channels_table[channel_configuration];
+  if (!*channels)
+    return FALSE;
+
+  if (audio_object_type == 5) {
+    GST_LOG_OBJECT (aacparse,
+        "Audio object type 5, so rereading sampling rate...");
+    if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
+      return FALSE;
+  }
+
+  GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
+      *sample_rate, *channels);
+
+  /* There's LOTS of stuff next, but we ignore it for now as we have
+     what we want (sample rate and number of channels */
+  GST_DEBUG_OBJECT (aacparse,
+      "Need more code to parse humongous LOAS data, currently ignored");
+  if (bits)
+    *bits = 0;
+  return TRUE;
+}
+
+
+static gboolean
+gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
+    guint avail, gint * sample_rate, gint * channels, gint * version)
+{
+  GstBitReader br;
+  guint8 u8, v, vA;
+
+  /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
+  if (version)
+    *version = 4;
+
+  gst_bit_reader_init (&br, data, avail);
+
+  /* skip sync word (11 bits) and size (13 bits) */
+  gst_bit_reader_skip (&br, 11 + 13);
+
+  /* First bit is "use last config" */
+  if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
+    return FALSE;
+  if (u8) {
+    GST_DEBUG_OBJECT (aacparse, "Frame uses previous config");
+    if (!aacparse->sample_rate || !aacparse->channels) {
+      GST_WARNING_OBJECT (aacparse, "No previous config to use");
+    }
+    *sample_rate = aacparse->sample_rate;
+    *channels = aacparse->channels;
+    return TRUE;
+  }
+
+  GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
+
+  if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
+    return FALSE;
+  if (v) {
+    if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
+      return FALSE;
+  } else
+    vA = 0;
+
+  GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
+  if (vA == 0) {
+    guint8 same_time, subframes, num_program, prog;
+    if (v == 1) {
+      guint32 value;
+      if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
+        return FALSE;
+    }
+    if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
+      return FALSE;
+    if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
+      return FALSE;
+    if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
+      return FALSE;
+    GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
+        same_time, subframes, num_program);
+
+    for (prog = 0; prog <= num_program; ++prog) {
+      guint8 num_layer, layer;
+      if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
+        return FALSE;
+      GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
+
+      for (layer = 0; layer <= num_layer; ++layer) {
+        guint8 use_same_config;
+        if (prog == 0 && layer == 0) {
+          use_same_config = 0;
+        } else {
+          if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
+            return FALSE;
+        }
+        if (!use_same_config) {
+          if (v == 0) {
+            if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
+                    sample_rate, channels, NULL))
+              return FALSE;
+          } else {
+            guint32 bits, asc_len;
+            if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
+              return FALSE;
+            if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
+                    sample_rate, channels, &bits))
+              return FALSE;
+            asc_len -= bits;
+            gst_bit_reader_skip (&br, asc_len);
+          }
+        }
+      }
+    }
+    GST_WARNING_OBJECT (aacparse, "More data ignored");
+  } else {
+    GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
+  }
+  return TRUE;
+}
+
+/**
+ * gst_aac_parse_loas_get_frame_len:
+ * @data: block of data containing a LOAS header.
+ *
+ * This function calculates LOAS frame length from the given header.
+ *
+ * Returns: size of the LOAS frame.
+ */
+static inline guint
+gst_aac_parse_loas_get_frame_len (const guint8 * data)
+{
+  return (((data[1] & 0x1f) << 8) | data[2]) + 3;
+}
+
+
+/**
+ * gst_aac_parse_check_loas_frame:
+ * @aacparse: #GstAacParse.
+ * @data: Data to be checked.
+ * @avail: Amount of data passed.
+ * @framesize: If valid LOAS frame was found, this will be set to tell the
+ *             found frame size in bytes.
+ * @needed_data: If frame was not found, this may be set to tell how much
+ *               more data is needed in the next round to detect the frame
+ *               reliably. This may happen when a frame header candidate
+ *               is found but it cannot be guaranteed to be the header without
+ *               peeking the following data.
+ *
+ * Check if the given data contains contains LOAS frame. The algorithm
+ * will examine LOAS frame header and calculate the frame size. Also, another
+ * consecutive LOAS frame header need to be present after the found frame.
+ * Otherwise the data is not considered as a valid LOAS frame. However, this
+ * "extra check" is omitted when EOS has been received. In this case it is
+ * enough when data[0] contains a valid LOAS header.
+ *
+ * This function may set the #needed_data to indicate that a possible frame
+ * candidate has been found, but more data (#needed_data bytes) is needed to
+ * be absolutely sure. When this situation occurs, FALSE will be returned.
+ *
+ * When a valid frame is detected, this function will use
+ * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
+ * to set the needed bytes for next frame.This way next data chunk is already
+ * of correct size.
+ *
+ * LOAS can have three different formats, if I read the spec correctly. Only
+ * one of them is supported here, as the two samples I have use this one.
+ *
+ * Returns: TRUE if the given data contains a valid LOAS header.
+ */
+static gboolean
+gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
+    const guint8 * data, const guint avail, gboolean drain,
+    guint * framesize, guint * needed_data)
+{
+  *needed_data = 0;
+
+  /* 3 byte header */
+  if (G_UNLIKELY (avail < 3))
+    return FALSE;
+
+  if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
+    *framesize = gst_aac_parse_loas_get_frame_len (data);
+    GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize);
+
+    /* In EOS mode this is enough. No need to examine the data further.
+       We also relax the check when we have sync, on the assumption that
+       if we're not looking at random data, we have a much higher chance
+       to get the correct sync, and this avoids losing two frames when
+       a single bit corruption happens. */
+    if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
+      return TRUE;
+    }
+
+    if (*framesize + LOAS_MAX_SIZE > avail) {
+      /* We have found a possible frame header candidate, but can't be
+         sure since we don't have enough data to check the next frame */
+      GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
+          *framesize + LOAS_MAX_SIZE, avail);
+      *needed_data = *framesize + LOAS_MAX_SIZE;
+      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
+          *framesize + LOAS_MAX_SIZE);
+      return FALSE;
+    }
+
+    if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
+      guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
+
+      GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
+      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
+          nextlen + LOAS_MAX_SIZE);
+      return TRUE;
+    }
+  }
+  return FALSE;
+}
+
 /* caller ensure sufficient data */
 static inline void
 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
@@ -412,7 +715,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
     guint * framesize, gint * skipsize)
 {
   gboolean found = FALSE;
-  guint need_data = 0;
+  guint need_data_adts = 0, need_data_loas;
   guint i = 0;
 
   GST_DEBUG_OBJECT (aacparse, "Parsing header data");
@@ -421,12 +724,16 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
      stream */
 
   /* Can we even parse the header? */
-  if (avail < ADTS_MAX_SIZE)
+  if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
+    GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
     return FALSE;
+  }
 
   for (i = 0; i < avail - 4; i++) {
     if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
+        ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) ||
         strncmp ((char *) data + i, "ADIF", 4) == 0) {
+      GST_DEBUG_OBJECT (aacparse, "Found ADIF signature at offset %u", i);
       found = TRUE;
 
       if (i) {
@@ -446,7 +753,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
   }
 
   if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
-          framesize, &need_data)) {
+          framesize, &need_data_adts)) {
     gint rate, channels;
 
     GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
@@ -464,7 +771,38 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
     gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
 
     return TRUE;
-  } else if (need_data) {
+  }
+
+  if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
+          framesize, &need_data_loas)) {
+    gint rate, channels;
+
+    GST_INFO ("LOAS, framesize: %d", *framesize);
+
+    aacparse->header_type = DSPAAC_HEADER_LOAS;
+
+    if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
+            &channels, &aacparse->mpegversion)) {
+      GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
+      return FALSE;
+    }
+
+    if (rate && channels) {
+      gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
+          aacparse->frame_samples, 2, 2);
+
+      GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
+          rate, channels, aacparse->object_type, aacparse->mpegversion);
+      aacparse->sample_rate = rate;
+      aacparse->channels = channels;
+    }
+
+    gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
+
+    return TRUE;
+  }
+
+  if (need_data_adts || need_data_loas) {
     /* This tells the parent class not to skip any data */
     *skipsize = 0;
     return FALSE;
@@ -612,6 +950,18 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
           needed_data);
     }
 
+  } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
+    guint needed_data = 1024;
+
+    ret = gst_aac_parse_check_loas_frame (aacparse, data,
+        size, GST_BASE_PARSE_DRAINING (parse), framesize, &needed_data);
+
+    if (!ret) {
+      GST_DEBUG ("buffer didn't contain valid frame");
+      gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
+          needed_data);
+    }
+
   } else {
     GST_DEBUG ("buffer didn't contain valid frame");
     gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
@@ -632,7 +982,7 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
  *
  * Also determines frame overhead.
  * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
- * a per-frame header.
+ * a per-frame header. LOAS has 3 bytes.
  *
  * We're making a couple of simplifying assumptions:
  *
@@ -659,36 +1009,70 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
   aacparse = GST_AAC_PARSE (parse);
   buffer = frame->buffer;
 
-  if (G_UNLIKELY (aacparse->header_type != DSPAAC_HEADER_ADTS))
-    return ret;
+  if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
+    /* see above */
+    frame->overhead = 7;
 
-  /* see above */
-  frame->overhead = 7;
+    data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+    gst_aac_parse_parse_adts_header (aacparse, data,
+        &rate, &channels, NULL, NULL);
+    gst_buffer_unmap (buffer, data, size);
 
-  data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
-  gst_aac_parse_parse_adts_header (aacparse, data,
-      &rate, &channels, NULL, NULL);
-  gst_buffer_unmap (buffer, data, size);
-
-  GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
+    GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
 
-  if (G_UNLIKELY (rate != aacparse->sample_rate
-          || channels != aacparse->channels)) {
-    GstCaps *sinkcaps;
+    if (G_UNLIKELY (rate != aacparse->sample_rate
+            || channels != aacparse->channels)) {
+      GstCaps *sinkcaps;
 
-    aacparse->sample_rate = rate;
-    aacparse->channels = channels;
+      aacparse->sample_rate = rate;
+      aacparse->channels = channels;
 
-    sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE (aacparse)->sinkpad);
-    if (!gst_aac_parse_set_src_caps (aacparse, sinkcaps)) {
-      /* If linking fails, we need to return appropriate error */
+      sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE (aacparse)->sinkpad);
+      if (!gst_aac_parse_set_src_caps (aacparse, sinkcaps)) {
+        /* If linking fails, we need to return appropriate error */
+        ret = GST_FLOW_NOT_LINKED;
+      }
       gst_caps_unref (sinkcaps);
-      ret = GST_FLOW_NOT_LINKED;
+
+      gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
+          aacparse->sample_rate, aacparse->frame_samples, 2, 2);
     }
-    gst_caps_unref (sinkcaps);
+  } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
+    gboolean setcaps = FALSE;
+
+    /* see above */
+    frame->overhead = 3;
+
+    data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+    if (!gst_aac_parse_read_loas_config (aacparse, data, size, &rate, &channels,
+            NULL)) {
+      GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
+    } else if (G_UNLIKELY (rate != aacparse->sample_rate
+            || channels != aacparse->channels)) {
+      aacparse->sample_rate = rate;
+      aacparse->channels = channels;
+      setcaps = TRUE;
+      GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
+          channels);
+    }
+    gst_buffer_unmap (buffer, data, size);
+
+    /* We want to set caps both at start, and when rate/channels change.
+       Since only some LOAS frames have that info, we may receive frames
+       before knowing about rate/channels. */
+    if (setcaps
+        || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
+      GstCaps *sinkcaps =
+          gst_pad_get_current_caps (GST_BASE_PARSE (aacparse)->sinkpad);
+      if (!gst_aac_parse_set_src_caps (aacparse, sinkcaps)) {
+        /* If linking fails, we need to return appropriate error */
+        ret = GST_FLOW_NOT_LINKED;
+      }
+      gst_caps_unref (sinkcaps);
 
-    gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
-        aacparse->sample_rate, aacparse->frame_samples, 2, 2);
+      gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
+          aacparse->sample_rate, aacparse->frame_samples, 2, 2);
+    }
   }
 
   return ret;
diff --git a/gst/audioparsers/gstaacparse.h b/gst/audioparsers/gstaacparse.h
index 1907c2e44..11f75e683 100644
--- a/gst/audioparsers/gstaacparse.h
+++ b/gst/audioparsers/gstaacparse.h
@@ -45,6 +45,7 @@ G_BEGIN_DECLS
  * @DSPAAC_HEADER_UNKNOWN: Unknown (not recognized) header.
  * @DSPAAC_HEADER_ADIF: ADIF header found.
  * @DSPAAC_HEADER_ADTS: ADTS header found.
+ * @DSPAAC_HEADER_LOAS: LOAS header found.
  * @DSPAAC_HEADER_NONE: Raw stream, no header.
  *
  * Type header enumeration set in #header_type.
@@ -54,6 +55,7 @@ typedef enum {
   DSPAAC_HEADER_UNKNOWN,
   DSPAAC_HEADER_ADIF,
   DSPAAC_HEADER_ADTS,
+  DSPAAC_HEADER_LOAS,
   DSPAAC_HEADER_NONE
 } GstAacHeaderType;
 
diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c
index 26b619585..1c4b85326 100644
--- a/gst/audioparsers/gstac3parse.c
+++ b/gst/audioparsers/gstac3parse.c
@@ -324,7 +324,9 @@ gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf,
 
   /* spec not quite clear here: decoder should decode if less than 8,
    * but seemingly only defines 6 and 8 cases */
-  if (bsid > 8) {
+  /* Files with 9 and 10 happen, and seem to comply with the <= 8
+     format, so let them through. The spec says nothing about 9 and 10 */
+  if (bsid > 10) {
     GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid);
     goto cleanup;
   } else if (bsid != 8 && bsid != 6) {
diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c
index da0943af7..1c2384b0b 100644
--- a/gst/avi/gstavidemux.c
+++ b/gst/avi/gstavidemux.c
@@ -45,6 +45,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <stdio.h>
 
diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c
index bea410e66..f34046e60 100644
--- a/gst/avi/gstavimux.c
+++ b/gst/avi/gstavimux.c
@@ -190,10 +190,10 @@ static GstStaticPadTemplate audio_sink_factory =
 
 static void gst_avi_mux_pad_reset (GstAviPad * avipad, gboolean free);
 
-static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads * pads,
+static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads2 * pads,
     GstAviMux * avimux);
-static gboolean gst_avi_mux_handle_event (GstPad * pad, GstObject * parent,
-    GstEvent * event);
+static gboolean gst_avi_mux_handle_event (GstCollectPads2 * pad,
+    GstCollectData2 * data, GstEvent * event, gpointer user_data);
 static GstPad *gst_avi_mux_request_new_pad (GstElement * element,
     GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
 static void gst_avi_mux_release_pad (GstElement * element, GstPad * pad);
@@ -378,10 +378,13 @@ gst_avi_mux_init (GstAviMux * avimux)
   /* property */
   avimux->enable_large_avi = DEFAULT_BIGFILE;
 
-  avimux->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (avimux->collect,
-      (GstCollectPadsFunction) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)),
+  avimux->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_function (avimux->collect,
+      (GstCollectPads2Function) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)),
       avimux);
+  gst_collect_pads2_set_event_function (avimux->collect,
+      (GstCollectPads2EventFunction) (GST_DEBUG_FUNCPTR
+          (gst_avi_mux_handle_event)), avimux);
 
   /* set to clean state */
   gst_avi_mux_reset (avimux);
@@ -977,15 +980,9 @@ gst_avi_mux_request_new_pad (GstElement * element,
 
   g_free (name);
 
-  avipad->collect = gst_collect_pads_add_pad (avimux->collect,
-      newpad, sizeof (GstAviCollectData), NULL);
+  avipad->collect = gst_collect_pads2_add_pad (avimux->collect,
+      newpad, sizeof (GstAviCollectData));
   ((GstAviCollectData *) (avipad->collect))->avipad = avipad;
-  /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
-   * element no access to events */
-  avimux->collect_event = GST_PAD_EVENTFUNC (newpad);
-  gst_pad_set_event_function (newpad,
-      GST_DEBUG_FUNCPTR (gst_avi_mux_handle_event));
 
   if (!gst_element_add_pad (element, newpad))
     goto pad_add_failed;
@@ -1038,7 +1035,7 @@ gst_avi_mux_release_pad (GstElement * element, GstPad * pad)
        * as it also represent number of streams present */
       avipad->collect = NULL;
       GST_DEBUG_OBJECT (avimux, "removed pad '%s'", GST_PAD_NAME (pad));
-      gst_collect_pads_remove_pad (avimux->collect, pad);
+      gst_collect_pads2_remove_pad (avimux->collect, pad);
       gst_element_remove_pad (element, pad);
       /* if not started yet, we can remove any sign this pad ever existed */
       /* in this case _start will take care of the real pad count */
@@ -1833,12 +1830,13 @@ gst_avi_mux_restart_file (GstAviMux * avimux)
 
 /* handle events (search) */
 static gboolean
-gst_avi_mux_handle_event (GstPad * pad, GstObject * parent, GstEvent * event)
+gst_avi_mux_handle_event (GstCollectPads2 * pads, GstCollectData2 * data,
+    GstEvent * event, gpointer user_data)
 {
   GstAviMux *avimux;
-  gboolean ret = TRUE;
+  gboolean ret = FALSE;
 
-  avimux = GST_AVI_MUX (parent);
+  avimux = GST_AVI_MUX (user_data);
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CAPS:
@@ -1850,15 +1848,15 @@ gst_avi_mux_handle_event (GstPad * pad, GstObject * parent, GstEvent * event)
       gst_event_parse_caps (event, &caps);
 
       /* find stream data */
-      collect_pad = (GstAviCollectData *) gst_pad_get_element_private (pad);
+      collect_pad = (GstAviCollectData *) data;
       g_assert (collect_pad);
       avipad = (GstAviVideoPad *) collect_pad->avipad;
       g_assert (avipad);
 
       if (avipad->parent.is_video) {
-        ret = gst_avi_mux_vidsink_set_caps (pad, caps);
+        ret = gst_avi_mux_vidsink_set_caps (GST_PAD (avipad), caps);
       } else {
-        ret = gst_avi_mux_audsink_set_caps (pad, caps);
+        ret = gst_avi_mux_audsink_set_caps (GST_PAD (avipad), caps);
       }
       break;
     }
@@ -1875,10 +1873,7 @@ gst_avi_mux_handle_event (GstPad * pad, GstObject * parent, GstEvent * event)
       break;
   }
 
-  /* now GstCollectPads can take care of the rest, e.g. EOS */
-  if (ret)
-    ret = avimux->collect_event (pad, parent, event);
-
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
   return ret;
 }
 
@@ -1909,7 +1904,7 @@ gst_avi_mux_do_buffer (GstAviMux * avimux, GstAviPad * avipad)
   guint flags;
   gsize datasize;
 
-  data = gst_collect_pads_pop (avimux->collect, avipad->collect);
+  data = gst_collect_pads2_pop (avimux->collect, avipad->collect);
   /* arrange downstream running time */
   data = gst_buffer_make_writable (data);
   GST_BUFFER_TIMESTAMP (data) =
@@ -2033,7 +2028,7 @@ gst_avi_mux_do_one_buffer (GstAviMux * avimux)
     if (!avipad->hdr.fcc_handler)
       goto not_negotiated;
 
-    buffer = gst_collect_pads_peek (avimux->collect, avipad->collect);
+    buffer = gst_collect_pads2_peek (avimux->collect, avipad->collect);
     if (!buffer)
       continue;
     time = GST_BUFFER_TIMESTAMP (buffer);
@@ -2046,7 +2041,7 @@ gst_avi_mux_do_one_buffer (GstAviMux * avimux)
       if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) {
         GST_DEBUG_OBJECT (avimux, "clipping buffer on pad %s outside segment",
             GST_PAD_NAME (avipad->collect->pad));
-        buffer = gst_collect_pads_pop (avimux->collect, avipad->collect);
+        buffer = gst_collect_pads2_pop (avimux->collect, avipad->collect);
         gst_buffer_unref (buffer);
         return GST_FLOW_OK;
       }
@@ -2085,7 +2080,7 @@ not_negotiated:
 }
 
 static GstFlowReturn
-gst_avi_mux_collect_pads (GstCollectPads * pads, GstAviMux * avimux)
+gst_avi_mux_collect_pads (GstCollectPads2 * pads, GstAviMux * avimux)
 {
   GstFlowReturn res;
 
@@ -2144,12 +2139,12 @@ gst_avi_mux_change_state (GstElement * element, GstStateChange transition)
 
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_collect_pads_start (avimux->collect);
+      gst_collect_pads2_start (avimux->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_collect_pads_stop (avimux->collect);
+      gst_collect_pads2_stop (avimux->collect);
       break;
     default:
       break;
diff --git a/gst/avi/gstavimux.h b/gst/avi/gstavimux.h
index 0d903928e..cb0cdb7b3 100644
--- a/gst/avi/gstavimux.h
+++ b/gst/avi/gstavimux.h
@@ -23,7 +23,7 @@
 
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 #include <gst/riff/riff-ids.h>
 #include "avi-ids.h"
 
@@ -74,7 +74,7 @@ typedef GstFlowReturn (*GstAviPadHook) (GstAviMux * avi, GstAviPad * avipad,
 struct _GstAviPad {
   /* do not extend, link to it */
   /* is NULL if original sink request pad has been removed */
-  GstCollectData *collect;
+  GstCollectData2 *collect;
 
   /* type */
   gboolean is_video;
@@ -129,7 +129,7 @@ typedef struct _GstAviAudioPad {
 
 typedef struct _GstAviCollectData {
   /* extend the CollectData */
-  GstCollectData collect;
+  GstCollectData2 collect;
 
   GstAviPad      *avipad;
 } GstAviCollectData;
@@ -143,8 +143,7 @@ struct _GstAviMux {
   GSList              *sinkpads;
   /* video restricted to 1 pad */
   guint               video_pads, audio_pads;
-  GstCollectPads      *collect;
-  GstPadEventFunction  collect_event;
+  GstCollectPads2     *collect;
 
   /* the AVI header */
   /* still some single stream video data in mux struct */
diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c
index ee7ffee22..da7f8e43c 100644
--- a/gst/equalizer/gstiirequalizer.c
+++ b/gst/equalizer/gstiirequalizer.c
@@ -32,6 +32,8 @@
 #include "gstiirequalizer3bands.h"
 #include "gstiirequalizer10bands.h"
 
+#include "gst/glib-compat-private.h"
+
 GST_DEBUG_CATEGORY (equalizer_debug);
 #define GST_CAT_DEFAULT equalizer_debug
 
diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c
index 324c2189c..7603dbaf8 100644
--- a/gst/flv/gstflvdemux.c
+++ b/gst/flv/gstflvdemux.c
@@ -34,6 +34,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include "gstflvdemux.h"
 #include "gstflvmux.h"
 
diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c
index a7189e2ea..2249d3044 100644
--- a/gst/flv/gstflvmux.c
+++ b/gst/flv/gstflvmux.c
@@ -89,7 +89,11 @@ G_DEFINE_TYPE_WITH_CODE (GstFlvMux, gst_flv_mux, GST_TYPE_ELEMENT,
 
 static void gst_flv_mux_finalize (GObject * object);
 static GstFlowReturn
-gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data);
+gst_flv_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata,
+    GstBuffer * buf, gpointer user_data);
+static gboolean
+gst_flv_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data,
+    GstEvent * event, gpointer user_data);
 
 static gboolean gst_flv_mux_handle_src_event (GstPad * pad, GstObject * parent,
     GstEvent * event);
@@ -209,9 +213,13 @@ gst_flv_mux_init (GstFlvMux * mux)
 
   mux->new_tags = FALSE;
 
-  mux->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (mux->collect,
-      (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_flv_mux_collected), mux);
+  mux->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_buffer_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_flv_mux_handle_buffer), mux);
+  gst_collect_pads2_set_event_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_flv_mux_handle_sink_event), mux);
+  gst_collect_pads2_set_clip_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), mux);
 
   gst_flv_mux_reset (GST_ELEMENT (mux));
 }
@@ -273,11 +281,11 @@ gst_flv_mux_handle_src_event (GstPad * pad, GstObject * parent,
 }
 
 static gboolean
-gst_flv_mux_handle_sink_event (GstPad * pad, GstObject * parent,
-    GstEvent * event)
+gst_flv_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data,
+    GstEvent * event, gpointer user_data)
 {
-  GstFlvMux *mux = GST_FLV_MUX (parent);
-  gboolean ret = TRUE;
+  GstFlvMux *mux = GST_FLV_MUX (user_data);
+  gboolean ret = FALSE;
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CAPS:
@@ -288,13 +296,13 @@ gst_flv_mux_handle_sink_event (GstPad * pad, GstObject * parent,
       gst_event_parse_caps (event, &caps);
 
       /* find stream data */
-      flvpad = (GstFlvPad *) gst_pad_get_element_private (pad);
+      flvpad = (GstFlvPad *) data;
       g_assert (flvpad);
 
       if (flvpad->video) {
-        ret = gst_flv_mux_video_pad_setcaps (pad, caps);
+        ret = gst_flv_mux_video_pad_setcaps (data->pad, caps);
       } else {
-        ret = gst_flv_mux_audio_pad_setcaps (pad, caps);
+        ret = gst_flv_mux_audio_pad_setcaps (data->pad, caps);
       }
       /* and eat */
       ret = FALSE;
@@ -315,10 +323,7 @@ gst_flv_mux_handle_sink_event (GstPad * pad, GstObject * parent,
       break;
   }
 
-  /* now GstCollectPads can take care of the rest, e.g. EOS */
-  if (ret)
-    ret = mux->collect_event (pad, parent, event);
-
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
   return ret;
 }
 
@@ -567,20 +572,12 @@ gst_flv_mux_request_new_pad (GstElement * element,
 
   pad = gst_pad_new_from_template (templ, name);
   cpad = (GstFlvPad *)
-      gst_collect_pads_add_pad (mux->collect, pad, sizeof (GstFlvPad), NULL);
+      gst_collect_pads2_add_pad (mux->collect, pad, sizeof (GstFlvPad));
 
   cpad->audio_codec_data = NULL;
   cpad->video_codec_data = NULL;
   gst_flv_mux_reset_pad (mux, cpad, video);
 
-  /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
-   * element no access to events.
-   */
-  mux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (pad);
-  gst_pad_set_event_function (pad,
-      GST_DEBUG_FUNCPTR (gst_flv_mux_handle_sink_event));
-
   gst_pad_set_active (pad, TRUE);
   gst_element_add_pad (element, pad);
 
@@ -594,7 +591,7 @@ gst_flv_mux_release_pad (GstElement * element, GstPad * pad)
   GstFlvPad *cpad = (GstFlvPad *) gst_pad_get_element_private (pad);
 
   gst_flv_mux_reset_pad (mux, cpad, cpad->video);
-  gst_collect_pads_remove_pad (mux->collect, pad);
+  gst_collect_pads2_remove_pad (mux->collect, pad);
   gst_element_remove_pad (element, pad);
 }
 
@@ -677,7 +674,7 @@ gst_flv_mux_create_number_script_value (const gchar * name, gdouble value)
 }
 
 static GstBuffer *
-gst_flv_mux_create_metadata (GstFlvMux * mux)
+gst_flv_mux_create_metadata (GstFlvMux * mux, gboolean full)
 {
   const GstTagList *tags;
   GstBuffer *script_tag, *tmp;
@@ -719,6 +716,9 @@ gst_flv_mux_create_metadata (GstFlvMux * mux)
   GST_WRITE_UINT32_BE (data + 1, n_tags);
   script_tag = gst_buffer_join (script_tag, tmp);
 
+  if (!full)
+    goto tags;
+
   /* Some players expect the 'duration' to be always set. Fill it out later,
      after querying the pads or after getting EOS */
   if (!mux->streamable) {
@@ -739,6 +739,7 @@ gst_flv_mux_create_metadata (GstFlvMux * mux)
     GST_DEBUG_OBJECT (mux, "not preallocating index, streamable mode");
   }
 
+tags:
   for (i = 0; tags && i < n_tags; i++) {
     const gchar *tag_name =
         gst_structure_nth_field_name ((const GstStructure *) tags, i);
@@ -777,12 +778,15 @@ gst_flv_mux_create_metadata (GstFlvMux * mux)
     }
   }
 
+  if (!full)
+    goto end;
+
   if (mux->duration == GST_CLOCK_TIME_NONE) {
     GSList *l;
     guint64 dur;
 
     for (l = mux->collect->data; l; l = l->next) {
-      GstCollectData *cdata = l->data;
+      GstCollectData2 *cdata = l->data;
 
       if (gst_pad_peer_query_duration (cdata->pad, GST_FORMAT_TIME,
               (gint64 *) & dur) && dur != GST_CLOCK_TIME_NONE) {
@@ -958,6 +962,14 @@ gst_flv_mux_create_metadata (GstFlvMux * mux)
     tags_written++;
   }
 
+end:
+
+  if (!tags_written) {
+    gst_buffer_unref (script_tag);
+    script_tag = NULL;
+    goto exit;
+  }
+
   _gst_buffer_new_and_alloc (2 + 0 + 1, &tmp, &data);
   data[0] = 0;                  /* 0 byte size */
   data[1] = 0;
@@ -977,6 +989,7 @@ gst_flv_mux_create_metadata (GstFlvMux * mux)
   GST_WRITE_UINT32_BE (data + 11 + 13 + 1, tags_written);
   gst_buffer_unmap (script_tag, data, -1);
 
+exit:
   return script_tag;
 }
 
@@ -1125,7 +1138,7 @@ gst_flv_mux_write_header (GstFlvMux * mux)
   GstFlowReturn ret;
 
   header = gst_flv_mux_create_header (mux);
-  metadata = gst_flv_mux_create_metadata (mux);
+  metadata = gst_flv_mux_create_metadata (mux, TRUE);
   video_codec_data = NULL;
   audio_codec_data = NULL;
 
@@ -1228,18 +1241,12 @@ gst_flv_mux_update_index (GstFlvMux * mux, GstBuffer * buffer, GstFlvPad * cpad)
 }
 
 static GstFlowReturn
-gst_flv_mux_write_buffer (GstFlvMux * mux, GstFlvPad * cpad)
+gst_flv_mux_write_buffer (GstFlvMux * mux, GstFlvPad * cpad, GstBuffer * buffer)
 {
   GstBuffer *tag;
-  GstBuffer *buffer =
-      gst_collect_pads_pop (mux->collect, (GstCollectData *) cpad);
   GstFlowReturn ret;
 
-  /* arrange downstream running time */
-  buffer = gst_buffer_make_writable (buffer);
-  GST_BUFFER_TIMESTAMP (buffer) =
-      gst_segment_to_running_time (&cpad->collect.segment,
-      GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buffer));
+  /* clipping function arranged for running_time */
 
   if (!mux->streamable)
     gst_flv_mux_update_index (mux, buffer, cpad);
@@ -1276,12 +1283,6 @@ gst_flv_mux_determine_duration (GstFlvMux * mux)
     }
   }
 
-  if (duration == GST_CLOCK_TIME_NONE) {
-    GST_DEBUG_OBJECT (mux, "not able to determine duration "
-        "from pad timestamps, assuming 0");
-    return 0;
-  }
-
   return duration;
 }
 
@@ -1296,6 +1297,7 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux)
   guint32 index_len, allocate_size;
   guint32 i, index_skip;
   GstSegment segment;
+  GstClockTime dur;
 
   if (mux->streamable)
     return GST_FLOW_OK;
@@ -1309,9 +1311,12 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux)
     return GST_FLOW_OK;
   }
 
-  /* if we were not able to determine the duration before, set it now */
-  if (mux->duration == GST_CLOCK_TIME_NONE)
-    mux->duration = gst_flv_mux_determine_duration (mux);
+  /* determine duration now based on our own timestamping,
+   * so that it is likely many times better and consistent
+   * than whatever obtained by some query */
+  dur = gst_flv_mux_determine_duration (mux);
+  if (dur != GST_CLOCK_TIME_NONE)
+    mux->duration = dur;
 
   /* rewrite the duration tag */
   d = gst_guint64_to_gdouble (mux->duration);
@@ -1420,14 +1425,13 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux)
 }
 
 static GstFlowReturn
-gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data)
+gst_flv_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata,
+    GstBuffer * buffer, gpointer user_data)
 {
   GstFlvMux *mux = GST_FLV_MUX (user_data);
   GstFlvPad *best;
   GstClockTime best_time;
   GstFlowReturn ret;
-  GSList *sl;
-  gboolean eos = TRUE;
 
   if (mux->state == GST_FLV_MUX_STATE_HEADER) {
     GstSegment segment;
@@ -1450,51 +1454,18 @@ gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data)
   }
 
   if (mux->new_tags) {
-    GstBuffer *buf = gst_flv_mux_create_metadata (mux);
-    gst_flv_mux_push (mux, buf);
+    GstBuffer *buf = gst_flv_mux_create_metadata (mux, FALSE);
+    if (buf)
+      gst_flv_mux_push (mux, buf);
     mux->new_tags = FALSE;
   }
 
-
-  best = NULL;
-  best_time = GST_CLOCK_TIME_NONE;
-  for (sl = mux->collect->data; sl; sl = sl->next) {
-    GstFlvPad *cpad = sl->data;
-    GstBuffer *buffer = gst_collect_pads_peek (pads, (GstCollectData *) cpad);
-    GstClockTime time;
-
-    if (!buffer)
-      continue;
-
-    eos = FALSE;
-
-    time = GST_BUFFER_TIMESTAMP (buffer);
-    gst_buffer_unref (buffer);
-
-    /* Use buffers without valid timestamp first */
-    if (!GST_CLOCK_TIME_IS_VALID (time)) {
-      GST_WARNING_OBJECT (pads, "Buffer without valid timestamp");
-
-      best_time = cpad->last_timestamp;
-      best = cpad;
-      break;
-    }
-
-    time = gst_segment_to_running_time (&cpad->collect.segment,
-        GST_FORMAT_TIME, time);
-    if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) {
-      GST_DEBUG_OBJECT (mux, "clipping buffer on pad %s outside segment",
-          GST_PAD_NAME (cpad->collect.pad));
-      buffer = gst_collect_pads_pop (pads, (GstCollectData *) cpad);
-      gst_buffer_unref (buffer);
-      return GST_FLOW_OK;
-    }
-
-    if (best == NULL || (GST_CLOCK_TIME_IS_VALID (best_time)
-            && time < best_time)) {
-      best = cpad;
-      best_time = time;
-    }
+  best = (GstFlvPad *) cdata;
+  if (best) {
+    g_assert (buffer);
+    best_time = GST_BUFFER_TIMESTAMP (buffer);
+  } else {
+    best_time = GST_CLOCK_TIME_NONE;
   }
 
   /* The FLV timestamp is an int32 field. For non-live streams error out if a
@@ -1503,17 +1474,17 @@ gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data)
   if (!mux->streamable && GST_CLOCK_TIME_IS_VALID (best_time)
       && best_time / GST_MSECOND > G_MAXINT32) {
     GST_WARNING_OBJECT (mux, "Timestamp larger than FLV supports - EOS");
-    eos = TRUE;
+    gst_buffer_unref (buffer);
+    buffer = NULL;
+    best = NULL;
   }
 
-  if (!eos && best) {
-    return gst_flv_mux_write_buffer (mux, best);
-  } else if (eos) {
+  if (best) {
+    return gst_flv_mux_write_buffer (mux, best, buffer);
+  } else {
     gst_flv_mux_rewrite_header (mux);
     gst_pad_push_event (mux->srcpad, gst_event_new_eos ());
     return GST_FLOW_EOS;
-  } else {
-    return GST_FLOW_OK;
   }
 }
 
@@ -1565,12 +1536,12 @@ gst_flv_mux_change_state (GstElement * element, GstStateChange transition)
     case GST_STATE_CHANGE_NULL_TO_READY:
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_collect_pads_start (mux->collect);
+      gst_collect_pads2_start (mux->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_collect_pads_stop (mux->collect);
+      gst_collect_pads2_stop (mux->collect);
       break;
     default:
       break;
diff --git a/gst/flv/gstflvmux.h b/gst/flv/gstflvmux.h
index fff51b9b3..b0e10ea32 100644
--- a/gst/flv/gstflvmux.h
+++ b/gst/flv/gstflvmux.h
@@ -22,7 +22,7 @@
 #define __GST_FLV_MUX_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 G_BEGIN_DECLS
 
@@ -39,7 +39,7 @@ G_BEGIN_DECLS
 
 typedef struct
 {
-  GstCollectData collect;
+  GstCollectData2 collect;
 
   gboolean video;
 
@@ -65,11 +65,9 @@ typedef struct _GstFlvMux {
   GstElement     element;
 
   GstPad         *srcpad;
-  GstCollectPads *collect;
+  GstCollectPads2 *collect;
 
   /* <private> */
-  GstPadEventFunction collect_event;
-
   GstFlvMuxState state;
   gboolean have_audio;
   gboolean have_video;
diff --git a/gst/imagefreeze/gstimagefreeze.c b/gst/imagefreeze/gstimagefreeze.c
index f4c67c76f..03c425d33 100644
--- a/gst/imagefreeze/gstimagefreeze.c
+++ b/gst/imagefreeze/gstimagefreeze.c
@@ -41,6 +41,12 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
+#include <gst/glib-compat-private.h>
+
 #include "gstimagefreeze.h"
 
 static void gst_image_freeze_finalize (GObject * object);
diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c
index c2cc2db78..e7e0b668c 100644
--- a/gst/interleave/interleave.c
+++ b/gst/interleave/interleave.c
@@ -227,7 +227,7 @@ static gboolean gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps);
 
 static GstCaps *gst_interleave_sink_getcaps (GstPad * pad);
 
-static GstFlowReturn gst_interleave_collected (GstCollectPads * pads,
+static GstFlowReturn gst_interleave_collected (GstCollectPads2 * pads,
     GstInterleave * self);
 
 static void
@@ -407,9 +407,9 @@ gst_interleave_init (GstInterleave * self, GstInterleaveClass * klass)
 
   gst_element_add_pad (GST_ELEMENT (self), self->src);
 
-  self->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (self->collect,
-      (GstCollectPadsFunction) gst_interleave_collected, self);
+  self->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_function (self->collect,
+      (GstCollectPads2Function) gst_interleave_collected, self);
 
   self->input_channel_positions = g_value_array_new (0);
   self->channel_positions_from_input = TRUE;
@@ -500,11 +500,10 @@ gst_interleave_request_new_pad (GstElement * element, GstPadTemplate * templ,
   gst_pad_set_getcaps_function (new_pad,
       GST_DEBUG_FUNCPTR (gst_interleave_sink_getcaps));
 
-  gst_collect_pads_add_pad (self->collect, new_pad, sizeof (GstCollectData),
-      NULL);
+  gst_collect_pads2_add_pad (self->collect, new_pad, sizeof (GstCollectData2));
 
   /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
+   * GstCollectPads2; because it sets its own event function giving the
    * element no access to events */
   self->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (new_pad);
   gst_pad_set_event_function (new_pad,
@@ -550,7 +549,7 @@ not_sink_pad:
 could_not_add:
   {
     GST_DEBUG_OBJECT (self, "could not add pad %s", GST_PAD_NAME (new_pad));
-    gst_collect_pads_remove_pad (self->collect, new_pad);
+    gst_collect_pads2_remove_pad (self->collect, new_pad);
     gst_object_unref (new_pad);
     return NULL;
   }
@@ -604,7 +603,7 @@ gst_interleave_release_pad (GstElement * element, GstPad * pad)
 
   GST_OBJECT_UNLOCK (self->collect);
 
-  gst_collect_pads_remove_pad (self->collect, pad);
+  gst_collect_pads2_remove_pad (self->collect, pad);
   gst_element_remove_pad (element, pad);
 }
 
@@ -626,7 +625,7 @@ gst_interleave_change_state (GstElement * element, GstStateChange transition)
       self->segment_position = 0;
       self->segment_rate = 1.0;
       gst_segment_init (&self->segment, GST_FORMAT_UNDEFINED);
-      gst_collect_pads_start (self->collect);
+      gst_collect_pads2_start (self->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
@@ -635,11 +634,11 @@ gst_interleave_change_state (GstElement * element, GstStateChange transition)
   }
 
   /* Stop before calling the parent's state change function as
-   * GstCollectPads might take locks and we would deadlock in that
+   * GstCollectPads2 might take locks and we would deadlock in that
    * case
    */
   if (transition == GST_STATE_CHANGE_PAUSED_TO_READY)
-    gst_collect_pads_stop (self->collect);
+    gst_collect_pads2_stop (self->collect);
 
   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
 
@@ -879,7 +878,7 @@ gst_interleave_sink_event (GstPad * pad, GstEvent * event)
       break;
   }
 
-  /* now GstCollectPads can take care of the rest, e.g. EOS */
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
   ret = self->collect_event (pad, event);
 
   gst_object_unref (self);
@@ -1160,7 +1159,7 @@ gst_interleave_src_event (GstPad * pad, GstEvent * event)
       /* check if we are flushing */
       if (flags & GST_SEEK_FLAG_FLUSH) {
         /* make sure we accept nothing anymore and return WRONG_STATE */
-        gst_collect_pads_set_flushing (self->collect, TRUE);
+        gst_collect_pads2_set_flushing (self->collect, TRUE);
 
         /* flushing seek, start flush downstream, the flush will be done
          * when all pads received a FLUSH_STOP. */
@@ -1195,7 +1194,7 @@ gst_interleave_src_event (GstPad * pad, GstEvent * event)
 }
 
 static GstFlowReturn
-gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
+gst_interleave_collected (GstCollectPads2 * pads, GstInterleave * self)
 {
   guint size;
   GstBuffer *outbuf;
@@ -1211,7 +1210,7 @@ gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
   g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
   g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED);
 
-  size = gst_collect_pads_available (pads);
+  size = gst_collect_pads2_available (pads);
 
   g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR);
 
@@ -1238,13 +1237,13 @@ gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
   memset (GST_BUFFER_DATA (outbuf), 0, size * self->channels);
 
   for (collected = pads->data; collected != NULL; collected = collected->next) {
-    GstCollectData *cdata;
+    GstCollectData2 *cdata;
     GstBuffer *inbuf;
     guint8 *outdata;
 
-    cdata = (GstCollectData *) collected->data;
+    cdata = (GstCollectData2 *) collected->data;
 
-    inbuf = gst_collect_pads_take_buffer (pads, cdata, size);
+    inbuf = gst_collect_pads2_take_buffer (pads, cdata, size);
     if (inbuf == NULL) {
       GST_DEBUG_OBJECT (cdata->pad, "No buffer available");
       goto next;
diff --git a/gst/interleave/interleave.h b/gst/interleave/interleave.h
index fb3b27415..abe3439e4 100644
--- a/gst/interleave/interleave.h
+++ b/gst/interleave/interleave.h
@@ -27,7 +27,7 @@
 #define __INTERLEAVE_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 G_BEGIN_DECLS
 
@@ -49,7 +49,7 @@ struct _GstInterleave
   GstElement element;
 
   /*< private >*/
-  GstCollectPads *collect;
+  GstCollectPads2 *collect;
 
   gint channels;
   gint padcounter;
diff --git a/gst/isomp4/gstqtmoovrecover.c b/gst/isomp4/gstqtmoovrecover.c
index 678ab3bb4..3fbfec329 100644
--- a/gst/isomp4/gstqtmoovrecover.c
+++ b/gst/isomp4/gstqtmoovrecover.c
@@ -65,6 +65,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <glib/gstdio.h>
 #include <gst/gst.h>
 
diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c
index 0c7caf6f4..510747827 100644
--- a/gst/isomp4/gstqtmux.c
+++ b/gst/isomp4/gstqtmux.c
@@ -226,11 +226,11 @@ static GstPad *gst_qt_mux_request_new_pad (GstElement * element,
 static void gst_qt_mux_release_pad (GstElement * element, GstPad * pad);
 
 /* event */
-static gboolean gst_qt_mux_sink_event (GstPad * pad, GstObject * parent,
-    GstEvent * event);
+static gboolean gst_qt_mux_sink_event (GstCollectPads2 * pads,
+    GstCollectData2 * data, GstEvent * event, gpointer user_data);
 
-static GstFlowReturn gst_qt_mux_collected (GstCollectPads * pads,
-    gpointer user_data);
+static GstFlowReturn gst_qt_mux_handle_buffer (GstCollectPads2 * pads,
+    GstCollectData2 * cdata, GstBuffer * buf, gpointer user_data);
 static GstFlowReturn gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad,
     GstBuffer * buf);
 
@@ -482,9 +482,13 @@ gst_qt_mux_init (GstQTMux * qtmux, GstQTMuxClass * qtmux_klass)
   gst_element_add_pad (GST_ELEMENT (qtmux), qtmux->srcpad);
 
   qtmux->sinkpads = NULL;
-  qtmux->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (qtmux->collect,
-      (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_qt_mux_collected), qtmux);
+  qtmux->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_buffer_function (qtmux->collect,
+      GST_DEBUG_FUNCPTR (gst_qt_mux_handle_buffer), qtmux);
+  gst_collect_pads2_set_event_function (qtmux->collect,
+      GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event), qtmux);
+  gst_collect_pads2_set_clip_function (qtmux->collect,
+      GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), qtmux);
 
   /* properties set to default upon construction */
 
@@ -1661,7 +1665,7 @@ gst_qt_mux_start_file (GstQTMux * qtmux)
         gst_buffer_unref (prefix);
 
       for (walk = qtmux->sinkpads; walk && !fail; walk = g_slist_next (walk)) {
-        GstCollectData *cdata = (GstCollectData *) walk->data;
+        GstCollectData2 *cdata = (GstCollectData2 *) walk->data;
         GstQTPad *qpad = (GstQTPad *) cdata;
         /* write info for each stream */
         fail = atoms_recov_write_trak_info (qtmux->moov_recov_file, qpad->trak);
@@ -1761,7 +1765,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux)
 
   /* pushing last buffers for each pad */
   for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) {
-    GstCollectData *cdata = (GstCollectData *) walk->data;
+    GstCollectData2 *cdata = (GstCollectData2 *) walk->data;
     GstQTPad *qtpad = (GstQTPad *) cdata;
 
     /* avoid add_buffer complaining if not negotiated
@@ -1853,7 +1857,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux)
   /* check for late streams */
   first_ts = GST_CLOCK_TIME_NONE;
   for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) {
-    GstCollectData *cdata = (GstCollectData *) walk->data;
+    GstCollectData2 *cdata = (GstCollectData2 *) walk->data;
     GstQTPad *qtpad = (GstQTPad *) cdata;
 
     if (!GST_CLOCK_TIME_IS_VALID (first_ts) ||
@@ -1866,7 +1870,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux)
       GST_TIME_ARGS (first_ts));
   /* add EDTSs for late streams */
   for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) {
-    GstCollectData *cdata = (GstCollectData *) walk->data;
+    GstCollectData2 *cdata = (GstCollectData2 *) walk->data;
     GstQTPad *qtpad = (GstQTPad *) cdata;
     guint32 lateness;
     guint32 duration;
@@ -2520,14 +2524,13 @@ not_negotiated:
 }
 
 static GstFlowReturn
-gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data)
+gst_qt_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata,
+    GstBuffer * buf, gpointer user_data)
 {
   GstFlowReturn ret = GST_FLOW_OK;
   GstQTMux *qtmux = GST_QT_MUX_CAST (user_data);
-  GSList *walk;
   GstQTPad *best_pad = NULL;
-  GstClockTime time, best_time = GST_CLOCK_TIME_NONE;
-  GstBuffer *buf;
+  GstClockTime best_time = GST_CLOCK_TIME_NONE;
 
   if (G_UNLIKELY (qtmux->state == GST_QT_MUX_STATE_STARTED)) {
     if ((ret = gst_qt_mux_start_file (qtmux)) != GST_FLOW_OK)
@@ -2539,52 +2542,14 @@ gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data)
   if (G_UNLIKELY (qtmux->state == GST_QT_MUX_STATE_EOS))
     return GST_FLOW_EOS;
 
-  /* select the best buffer */
-  walk = qtmux->collect->data;
-  while (walk) {
-    GstQTPad *pad;
-    GstCollectData *data;
-
-    data = (GstCollectData *) walk->data;
-    pad = (GstQTPad *) data;
-
-    walk = g_slist_next (walk);
-
-    buf = gst_collect_pads_peek (pads, data);
-    if (buf == NULL) {
-      GST_LOG_OBJECT (qtmux, "Pad %s has no buffers",
-          GST_PAD_NAME (pad->collect.pad));
-      continue;
-    }
-    time = GST_BUFFER_TIMESTAMP (buf);
-    gst_buffer_unref (buf);
-
-    /* invalid should pass */
-    if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (time))) {
-      time =
-          gst_segment_to_running_time (&data->segment, GST_FORMAT_TIME, time);
-      if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) {
-        GST_DEBUG_OBJECT (qtmux, "clipping buffer on pad %s outside segment",
-            GST_PAD_NAME (data->pad));
-        buf = gst_collect_pads_pop (pads, data);
-        gst_buffer_unref (buf);
-        return GST_FLOW_OK;
-      }
-    }
-
-    if (best_pad == NULL || !GST_CLOCK_TIME_IS_VALID (time) ||
-        (GST_CLOCK_TIME_IS_VALID (best_time) && time < best_time)) {
-      best_pad = pad;
-      best_time = time;
-    }
-  }
+  best_pad = (GstQTPad *) cdata;
 
+  /* clipping already converted to running time */
   if (best_pad != NULL) {
+    g_assert (buf);
+    best_time = GST_BUFFER_TIMESTAMP (buf);
     GST_LOG_OBJECT (qtmux, "selected pad %s with time %" GST_TIME_FORMAT,
         GST_PAD_NAME (best_pad->collect.pad), GST_TIME_ARGS (best_time));
-    buf = gst_collect_pads_pop (pads, &best_pad->collect);
-    buf = gst_buffer_make_writable (buf);
-    GST_BUFFER_TIMESTAMP (buf) = best_time;
     ret = gst_qt_mux_add_buffer (qtmux, best_pad, buf);
   } else {
     ret = gst_qt_mux_stop_file (qtmux);
@@ -3284,14 +3249,14 @@ refuse_renegotiation:
 }
 
 static gboolean
-gst_qt_mux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
+gst_qt_mux_sink_event (GstCollectPads2 * pads, GstCollectData2 * data,
+    GstEvent * event, gpointer user_data)
 {
-  gboolean ret;
   GstQTMux *qtmux;
   guint32 avg_bitrate = 0, max_bitrate = 0;
+  GstPad *pad = data->pad;
 
-  qtmux = GST_QT_MUX_CAST (parent);
-
+  qtmux = GST_QT_MUX_CAST (user_data);
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CAPS:
     {
@@ -3316,8 +3281,9 @@ gst_qt_mux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
       GST_OBJECT_LOCK (qtmux);
       mode = gst_tag_setter_get_tag_merge_mode (setter);
 
-      GST_DEBUG_OBJECT (qtmux, "received tag event");
       gst_event_parse_tag (event, &list);
+      GST_DEBUG_OBJECT (qtmux, "received tag event on pad %s:%s : %"
+          GST_PTR_FORMAT, GST_DEBUG_PAD_NAME (pad), list);
 
       gst_tag_setter_merge_tags (setter, list, mode);
       GST_OBJECT_UNLOCK (qtmux);
@@ -3339,9 +3305,8 @@ gst_qt_mux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
       break;
   }
 
-  ret = qtmux->collect_event (pad, parent, event);
-
-  return ret;
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
+  return FALSE;
 }
 
 static void
@@ -3363,7 +3328,7 @@ gst_qt_mux_release_pad (GstElement * element, GstPad * pad)
     }
   }
 
-  gst_collect_pads_remove_pad (mux->collect, pad);
+  gst_collect_pads2_remove_pad (mux->collect, pad);
 }
 
 static GstPad *
@@ -3407,8 +3372,8 @@ gst_qt_mux_request_new_pad (GstElement * element,
   newpad = gst_pad_new_from_template (templ, name);
   g_free (name);
   collect_pad = (GstQTPad *)
-      gst_collect_pads_add_pad (qtmux->collect, newpad, sizeof (GstQTPad),
-      (GstCollectDataDestroyNotify) (gst_qt_mux_pad_reset));
+      gst_collect_pads2_add_pad_full (qtmux->collect, newpad, sizeof (GstQTPad),
+      (GstCollectData2DestroyNotify) (gst_qt_mux_pad_reset), TRUE);
   /* set up pad */
   gst_qt_mux_pad_reset (collect_pad);
   collect_pad->trak = atom_trak_new (qtmux->context);
@@ -3422,14 +3387,6 @@ gst_qt_mux_request_new_pad (GstElement * element,
   else
     collect_pad->set_caps = GST_DEBUG_FUNCPTR (gst_qt_mux_video_sink_set_caps);
 
-  /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
-   * element no access to events.
-   */
-  qtmux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
-  gst_pad_set_event_function (newpad,
-      GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event));
-
   gst_pad_set_active (newpad, TRUE);
   gst_element_add_pad (element, newpad);
 
@@ -3566,13 +3523,13 @@ gst_qt_mux_change_state (GstElement * element, GstStateChange transition)
     case GST_STATE_CHANGE_NULL_TO_READY:
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_collect_pads_start (qtmux->collect);
+      gst_collect_pads2_start (qtmux->collect);
       qtmux->state = GST_QT_MUX_STATE_STARTED;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_collect_pads_stop (qtmux->collect);
+      gst_collect_pads2_stop (qtmux->collect);
       break;
     default:
       break;
diff --git a/gst/isomp4/gstqtmux.h b/gst/isomp4/gstqtmux.h
index fdbbb5a54..c17ff25de 100644
--- a/gst/isomp4/gstqtmux.h
+++ b/gst/isomp4/gstqtmux.h
@@ -44,7 +44,7 @@
 #define __GST_QT_MUX_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 #include "fourcc.h"
 #include "atoms.h"
@@ -82,7 +82,7 @@ typedef GstBuffer * (*GstQTPadPrepareBufferFunc) (GstQTPad * pad,
 
 struct _GstQTPad
 {
-  GstCollectData collect;       /* we extend the CollectData */
+  GstCollectData2 collect;       /* we extend the CollectData2 */
 
   /* fourcc id of stream */
   guint32 fourcc;
@@ -145,7 +145,7 @@ struct _GstQTMux
   GstElement element;
 
   GstPad *srcpad;
-  GstCollectPads *collect;
+  GstCollectPads2 *collect;
   GSList *sinkpads;
 
   /* state */
@@ -192,9 +192,6 @@ struct _GstQTMux
   guint32 fragment_duration;
   gboolean streamable;
 
-  /* for collect pads event handling function */
-  GstPadEventFunction collect_event;
-
   /* for request pad naming */
   guint video_pads, audio_pads;
 };
diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
index 2b9053fac..b7511c5c7 100644
--- a/gst/isomp4/qtdemux.c
+++ b/gst/isomp4/qtdemux.c
@@ -47,6 +47,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include "gst/gst-i18n-plugin.h"
 
 #include <glib/gprintf.h>
@@ -5317,8 +5321,13 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl)
   GST_LOG_OBJECT (qtdemux, "%u timestamp blocks", stream->n_sample_times);
 
   /* make sure there's enough data */
-  if (!qt_atom_parser_has_chunks (&stream->stts, stream->n_sample_times, 2 * 4))
-    goto corrupt_file;
+  if (!qt_atom_parser_has_chunks (&stream->stts, stream->n_sample_times, 8)) {
+    stream->n_sample_times = gst_byte_reader_get_remaining (&stream->stts) / 8;
+    GST_LOG_OBJECT (qtdemux, "overriding to %u timestamp blocks",
+        stream->n_sample_times);
+    if (!stream->n_sample_times)
+      goto corrupt_file;
+  }
 
   /* sync sample atom */
   stream->stps_present = FALSE;
diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c
index 404071c8c..f5fb27267 100644
--- a/gst/matroska/matroska-demux.c
+++ b/gst/matroska/matroska-demux.c
@@ -49,6 +49,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <math.h>
 #include <string.h>
 #include <glib/gprintf.h>
@@ -482,6 +486,8 @@ gst_matroska_demux_reset (GstElement * element)
     gst_buffer_unref (demux->common.cached_buffer);
     demux->common.cached_buffer = NULL;
   }
+
+  demux->invalid_duration = FALSE;
 }
 
 static GstBuffer *
@@ -1356,10 +1362,12 @@ gst_matroska_demux_query (GstMatroskaDemux * demux, GstPad * pad,
         GST_OBJECT_LOCK (demux);
         if (context)
           gst_query_set_position (query, GST_FORMAT_TIME,
-              context->pos - demux->stream_start_time);
+              MAX (context->pos, demux->stream_start_time) -
+              demux->stream_start_time);
         else
           gst_query_set_position (query, GST_FORMAT_TIME,
-              demux->common.segment.position - demux->stream_start_time);
+              MAX (demux->common.segment.position, demux->stream_start_time) -
+              demux->stream_start_time);
         GST_OBJECT_UNLOCK (demux);
       } else if (format == GST_FORMAT_DEFAULT && context
           && context->default_duration) {
@@ -1740,8 +1748,11 @@ gst_matroska_demux_search_pos (GstMatroskaDemux * demux, GstClockTime time)
   otime = demux->common.segment.position;
   GST_OBJECT_UNLOCK (demux);
 
+  /* sanitize */
+  time = MAX (time, demux->stream_start_time);
+
   /* avoid division by zero in first estimation below */
-  if (otime == 0)
+  if (otime <= demux->stream_start_time)
     otime = time;
 
 retry:
@@ -1918,6 +1929,14 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux,
    * segment when we close the current segment. */
   memcpy (&seeksegment, &demux->common.segment, sizeof (GstSegment));
 
+  /* pull mode without index means that the actual duration is not known,
+   * we might be playing a file that's still being recorded
+   * so, invalidate our current duration, which is only a moving target,
+   * and should not be used to clamp anything */
+  if (!demux->streaming && !demux->common.index && demux->invalid_duration) {
+    seeksegment.duration = GST_CLOCK_TIME_NONE;
+  }
+
   if (event) {
     GST_DEBUG_OBJECT (demux, "configuring seek");
     gst_segment_do_seek (&seeksegment, rate, format, flags,
@@ -1932,6 +1951,10 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux,
     }
   }
 
+  /* restore segment duration (if any effect),
+   * would be determined again when parsing, but anyway ... */
+  seeksegment.duration = demux->common.segment.duration;
+
   flush = ! !(flags & GST_SEEK_FLAG_FLUSH);
   keyunit = ! !(flags & GST_SEEK_FLAG_KEY_UNIT);
 
@@ -1952,7 +1975,7 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux,
               seeksegment.position, &demux->seek_index, &demux->seek_entry)) ==
       NULL) {
     /* pull mode without index can scan later on */
-    if (demux->common.index || demux->streaming) {
+    if (demux->streaming) {
       GST_DEBUG_OBJECT (demux, "No matching seek entry in index");
       GST_OBJECT_UNLOCK (demux);
       return FALSE;
@@ -1989,7 +2012,7 @@ next:
   GST_PAD_STREAM_LOCK (demux->common.sinkpad);
 
   /* pull mode without index can do some scanning */
-  if (!demux->streaming && !demux->common.index) {
+  if (!demux->streaming && !entry) {
     /* need to stop flushing upstream as we need it next */
     if (flush)
       gst_pad_push_event (demux->common.sinkpad,
@@ -2011,9 +2034,9 @@ next:
   if (keyunit) {
     GST_DEBUG_OBJECT (demux, "seek to key unit, adjusting segment start to %"
         GST_TIME_FORMAT, GST_TIME_ARGS (entry->time));
-    seeksegment.start = entry->time;
-    seeksegment.position = entry->time;
-    seeksegment.time = entry->time - demux->stream_start_time;
+    seeksegment.start = MAX (entry->time, demux->stream_start_time);
+    seeksegment.position = seeksegment.start;
+    seeksegment.time = seeksegment.start - demux->stream_start_time;
   }
 
 exit:
@@ -3347,9 +3370,9 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux,
       else if (GST_CLOCK_TIME_IS_VALID (segment->position))
         segment_duration = segment->position - segment->start;
       segment->base += segment_duration / fabs (segment->rate);
-      segment->start = lace_time;
+      segment->start = MAX (lace_time, demux->stream_start_time);
       segment->stop = GST_CLOCK_TIME_NONE;
-      segment->position = lace_time - demux->stream_start_time;
+      segment->position = segment->start - demux->stream_start_time;
       /* now convey our segment notion downstream */
       gst_matroska_demux_send_event (demux, gst_event_new_segment (segment));
       demux->need_segment = FALSE;
@@ -3535,14 +3558,15 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux,
 
         GST_OBJECT_LOCK (demux);
         if (demux->common.segment.duration == -1 ||
-            demux->common.segment.duration <
-            lace_time - demux->stream_start_time) {
+            demux->stream_start_time + demux->common.segment.duration <
+            last_stop_end) {
           demux->common.segment.duration =
               last_stop_end - demux->stream_start_time;
           GST_OBJECT_UNLOCK (demux);
           gst_element_post_message (GST_ELEMENT_CAST (demux),
               gst_message_new_duration (GST_OBJECT_CAST (demux),
                   GST_FORMAT_TIME, GST_CLOCK_TIME_NONE));
+          demux->invalid_duration = TRUE;
         } else {
           GST_OBJECT_UNLOCK (demux);
         }
@@ -4475,11 +4499,13 @@ pause:
       /* Close the segment, i.e. update segment stop with the duration
        * if no stop was set */
       if (GST_CLOCK_TIME_IS_VALID (demux->last_stop_end) &&
-          !GST_CLOCK_TIME_IS_VALID (demux->common.segment.stop)) {
+          !GST_CLOCK_TIME_IS_VALID (demux->common.segment.stop) &&
+          GST_CLOCK_TIME_IS_VALID (demux->common.segment.start) &&
+          demux->last_stop_end > demux->common.segment.start) {
         GstSegment segment = demux->common.segment;
         GstEvent *event;
 
-        segment.stop = MAX (demux->last_stop_end, segment.start);
+        segment.stop = demux->last_stop_end;
         event = gst_event_new_segment (&segment);
         gst_matroska_demux_send_event (demux, event);
       }
diff --git a/gst/matroska/matroska-demux.h b/gst/matroska/matroska-demux.h
index 65061c1d1..0a6312ea8 100644
--- a/gst/matroska/matroska-demux.h
+++ b/gst/matroska/matroska-demux.h
@@ -95,6 +95,9 @@ typedef struct _GstMatroskaDemux {
 
   /* gap handling */
   guint64                  max_gap_time;
+
+  /* for non-finalized files, with invalid segment duration */
+  gboolean                 invalid_duration;
 } GstMatroskaDemux;
 
 typedef struct _GstMatroskaDemuxClass {
diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c
index f02c171a1..5fe25f957 100644
--- a/gst/matroska/matroska-mux.c
+++ b/gst/matroska/matroska-mux.c
@@ -2,6 +2,7 @@
  * (c) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
  * (c) 2005 Michal Benes <michal.benes@xeris.cz>
  * (c) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * (c) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
  *
  * matroska-mux.c: matroska file/stream muxer
  *
@@ -183,10 +184,14 @@ static GstStaticPadTemplate audiosink_templ =
     );
 
 static GstStaticPadTemplate subtitlesink_templ =
-GST_STATIC_PAD_TEMPLATE ("subtitle_%u",
+    GST_STATIC_PAD_TEMPLATE ("subtitle_%d",
     GST_PAD_SINK,
     GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("subtitle/x-kate"));
+    GST_STATIC_CAPS ("subtitle/x-kate; "
+        "text/plain; application/x-ssa; application/x-ass; "
+        "application/x-usf; video/x-dvd-subpicture; "
+        "application/x-subtitle-unknown")
+    );
 
 static GArray *used_uids;
 G_LOCK_DEFINE_STATIC (used_uids);
@@ -199,8 +204,10 @@ G_DEFINE_TYPE_WITH_CODE (GstMatroskaMux, gst_matroska_mux, GST_TYPE_ELEMENT,
 static void gst_matroska_mux_finalize (GObject * object);
 
 /* Pads collected callback */
-static GstFlowReturn
-gst_matroska_mux_collected (GstCollectPads * pads, gpointer user_data);
+static GstFlowReturn gst_matroska_mux_handle_buffer (GstCollectPads2 * pads,
+    GstCollectData2 * data, GstBuffer * buf, gpointer user_data);
+static gboolean gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads,
+    GstCollectData2 * data, GstEvent * event, gpointer user_data);
 
 /* pad functions */
 static gboolean gst_matroska_mux_handle_src_event (GstPad * pad,
@@ -415,10 +422,13 @@ gst_matroska_mux_init (GstMatroskaMux * mux)
   gst_pad_set_event_function (mux->srcpad, gst_matroska_mux_handle_src_event);
   gst_element_add_pad (GST_ELEMENT (mux), mux->srcpad);
 
-  mux->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (mux->collect,
-      (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_matroska_mux_collected),
-      mux);
+  mux->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_clip_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), mux);
+  gst_collect_pads2_set_buffer_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_buffer), mux);
+  gst_collect_pads2_set_event_function (mux->collect,
+      GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_sink_event), mux);
 
   mux->ebml_write = gst_ebml_write_new (mux->srcpad);
   mux->doctype = GST_MATROSKA_DOCTYPE_MATROSKA;
@@ -535,12 +545,6 @@ gst_matroska_pad_reset (GstMatroskaPad * collect_pad, gboolean full)
     collect_pad->track = NULL;
   }
 
-  /* free cached buffer */
-  if (collect_pad->buffer != NULL) {
-    gst_buffer_unref (collect_pad->buffer);
-    collect_pad->buffer = NULL;
-  }
-
   if (!full && type != 0) {
     GstMatroskaTrackContext *context;
 
@@ -568,7 +572,6 @@ gst_matroska_pad_reset (GstMatroskaPad * collect_pad, gboolean full)
     /* TODO: check default values for the context */
     context->flags = GST_MATROSKA_TRACK_ENABLED | GST_MATROSKA_TRACK_DEFAULT;
     collect_pad->track = context;
-    collect_pad->buffer = NULL;
     collect_pad->duration = 0;
     collect_pad->start_ts = GST_CLOCK_TIME_NONE;
     collect_pad->end_ts = GST_CLOCK_TIME_NONE;
@@ -665,6 +668,44 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstObject * parent,
   return gst_pad_event_default (pad, parent, event);
 }
 
+static void
+gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context,
+    const guint * clut)
+{
+  gchar *clutv[17];
+  gchar *sclut;
+  gint i;
+  guint32 col;
+  gdouble y, u, v;
+  guint8 r, g, b;
+
+  /* produce comma-separated list in hex format */
+  for (i = 0; i < 16; ++i) {
+    col = clut[i];
+    /* replicate vobsub's slightly off RGB conversion calculation */
+    y = (((col >> 16) & 0xff) - 16) * 255 / 219;
+    u = ((col >> 8) & 0xff) - 128;
+    v = (col & 0xff) - 128;
+    r = CLAMP (1.0 * y + 1.4022 * u, 0, 255);
+    g = CLAMP (1.0 * y - 0.3456 * u - 0.7145 * v, 0, 255);
+    b = CLAMP (1.0 * y + 1.7710 * v, 0, 255);
+    clutv[i] = g_strdup_printf ("%02x%02x%02x", r, g, b);
+  }
+  clutv[i] = NULL;
+  sclut = g_strjoinv (",", clutv);
+
+  /* build codec private; only palette for now */
+  g_free (context->codec_priv);
+  context->codec_priv = (guint8 *) g_strdup_printf ("palette: %s", sclut);
+  /* include terminating 0 */
+  context->codec_priv_size = strlen ((gchar *) context->codec_priv) + 1;
+  g_free (sclut);
+  for (i = 0; i < 16; ++i) {
+    g_free (clutv[i]);
+  }
+}
+
+
 /**
  * gst_matroska_mux_handle_sink_event:
  * @pad: Pad which received the event.
@@ -675,14 +716,21 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstObject * parent,
  * Returns: #TRUE on success.
  */
 static gboolean
-gst_matroska_mux_handle_sink_event (GstPad * pad, GstObject * parent,
-    GstEvent * event)
+gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads,
+    GstCollectData2 * data, GstEvent * event, gpointer user_data)
 {
-  GstMatroskaTrackContext *context;
   GstMatroskaPad *collect_pad;
-  GstMatroskaMux *mux = GST_MATROSKA_MUX (parent);
+  GstMatroskaTrackContext *context;
+  GstMatroskaMux *mux;
+  GstPad *pad;
   GstTagList *list;
-  gboolean ret = TRUE;
+  gboolean ret = FALSE;
+
+  mux = GST_MATROSKA_MUX (user_data);
+  collect_pad = (GstMatroskaPad *) data;
+  pad = data->pad;
+  context = collect_pad->track;
+  g_assert (context);
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_CAPS:{
@@ -702,11 +750,6 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstObject * parent,
       GST_DEBUG_OBJECT (mux, "received tag event");
       gst_event_parse_tag (event, &list);
 
-      collect_pad = (GstMatroskaPad *) gst_pad_get_element_private (pad);
-      g_assert (collect_pad);
-      context = collect_pad->track;
-      g_assert (context);
-
       /* Matroska wants ISO 639-2B code, taglist most likely contains 639-1 */
       if (gst_tag_list_get_string (list, GST_TAG_LANGUAGE_CODE, &lang)) {
         const gchar *lang_code;
@@ -735,7 +778,6 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstObject * parent,
 
       gst_event_parse_segment (event, &segment);
       if (segment->format != GST_FORMAT_TIME) {
-        ret = FALSE;
         gst_event_unref (event);
         event = NULL;
       }
@@ -749,6 +791,31 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstObject * parent,
         gst_event_replace (&mux->force_key_unit_event, NULL);
         mux->force_key_unit_event = event;
         event = NULL;
+      } else if (gst_structure_has_name (structure, "application/x-gst-dvd") &&
+          !strcmp ("dvd-spu-clut-change",
+              gst_structure_get_string (structure, "event"))) {
+        gchar name[16];
+        gint i, value;
+        guint clut[16];
+
+        GST_DEBUG_OBJECT (pad, "New DVD colour table received");
+        if (context->type != GST_MATROSKA_TRACK_TYPE_SUBTITLE) {
+          GST_DEBUG_OBJECT (pad, "... discarding");
+          break;
+        }
+        /* first transform event data into table form */
+        for (i = 0; i < 16; i++) {
+          g_snprintf (name, sizeof (name), "clut%02d", i);
+          if (!gst_structure_get_int (structure, name, &value)) {
+            GST_ERROR_OBJECT (mux, "dvd-spu-clut-change event did not "
+                "contain %s field", name);
+            break;
+          }
+          clut[i] = value;
+        }
+
+        /* transform into private data for stream; text form */
+        gst_matroska_mux_build_vobsub_private (context, clut);
       }
       break;
     }
@@ -756,13 +823,19 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstObject * parent,
       break;
   }
 
-  /* now GstCollectPads can take care of the rest, e.g. EOS */
-  if (event)
-    ret = mux->collect_event (pad, parent, event);
-
+  /* now GstCollectPads2 can take care of the rest, e.g. EOS */
   return ret;
 }
 
+static void
+gst_matroska_mux_set_codec_id (GstMatroskaTrackContext * context,
+    const char *id)
+{
+  g_assert (context && id);
+  if (context->codec_id)
+    g_free (context->codec_id);
+  context->codec_id = g_strdup (id);
+}
 
 /**
  * gst_matroska_mux_video_pad_setcaps:
@@ -872,12 +945,13 @@ skip_details:
   /* find type */
   if (!strcmp (mimetype, "video/x-raw")) {
     const gchar *fstr;
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_UNCOMPRESSED);
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_VIDEO_UNCOMPRESSED);
     fstr = gst_structure_get_string (structure, "format");
     if (fstr && strlen (fstr) == 4)
       videocontext->fourcc = GST_STR_FOURCC (fstr);
   } else if (!strcmp (mimetype, "image/jpeg")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MJPEG);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_MJPEG);
   } else if (!strcmp (mimetype, "video/x-xvid") /* MS/VfW compatibility cases */
       ||!strcmp (mimetype, "video/x-huffyuv")
       || !strcmp (mimetype, "video/x-divx")
@@ -970,11 +1044,13 @@ skip_details:
           (guint8 *) bih + sizeof (gst_riff_strf_vids), -1);
     }
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC);
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC);
     context->codec_priv = (gpointer) bih;
     context->codec_priv_size = size;
   } else if (!strcmp (mimetype, "video/x-h264")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC);
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC);
 
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
@@ -991,7 +1067,7 @@ skip_details:
   } else if (!strcmp (mimetype, "video/x-theora")) {
     const GValue *streamheader;
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_THEORA);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_THEORA);
 
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
@@ -1006,22 +1082,25 @@ skip_details:
       goto refuse_caps;
     }
   } else if (!strcmp (mimetype, "video/x-dirac")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_DIRAC);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_DIRAC);
   } else if (!strcmp (mimetype, "video/x-vp8")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_VP8);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_VP8);
   } else if (!strcmp (mimetype, "video/mpeg")) {
     gint mpegversion;
 
     gst_structure_get_int (structure, "mpegversion", &mpegversion);
     switch (mpegversion) {
       case 1:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG1);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_MPEG1);
         break;
       case 2:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG2);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_MPEG2);
         break;
       case 4:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_ASP);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_ASP);
         break;
       default:
         goto refuse_caps;
@@ -1044,16 +1123,20 @@ skip_details:
     gst_structure_get_int (structure, "rmversion", &rmversion);
     switch (rmversion) {
       case 1:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO1);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO1);
         break;
       case 2:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO2);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO2);
         break;
       case 3:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO3);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO3);
         break;
       case 4:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO4);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO4);
         break;
       default:
         goto refuse_caps;
@@ -1551,13 +1634,16 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
 
         switch (layer) {
           case 1:
-            context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L1);
+            gst_matroska_mux_set_codec_id (context,
+                GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L1);
             break;
           case 2:
-            context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L2);
+            gst_matroska_mux_set_codec_id (context,
+                GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L2);
             break;
           case 3:
-            context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L3);
+            gst_matroska_mux_set_codec_id (context,
+                GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L3);
             break;
           default:
             goto refuse_caps;
@@ -1620,14 +1706,16 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
           goto refuse_caps;
         }
         if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&info))
-          context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_BE);
+          gst_matroska_mux_set_codec_id (context,
+              GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_BE);
         else
-          context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_LE);
+          gst_matroska_mux_set_codec_id (context,
+              GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_LE);
         break;
-
       case GST_AUDIO_FORMAT_F32LE:
       case GST_AUDIO_FORMAT_F64LE:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_FLOAT);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_AUDIO_PCM_FLOAT);
         break;
 
       default:
@@ -1636,11 +1724,10 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
     }
 
     audiocontext->bitdepth = GST_AUDIO_INFO_WIDTH (&info);
-
   } else if (!strcmp (mimetype, "audio/x-vorbis")) {
     const GValue *streamheader;
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_VORBIS);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_VORBIS);
 
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
@@ -1657,7 +1744,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
   } else if (!strcmp (mimetype, "audio/x-flac")) {
     const GValue *streamheader;
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_FLAC);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_FLAC);
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
       context->codec_priv = NULL;
@@ -1673,7 +1760,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
   } else if (!strcmp (mimetype, "audio/x-speex")) {
     const GValue *streamheader;
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_SPEEX);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_SPEEX);
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
       context->codec_priv = NULL;
@@ -1687,11 +1774,11 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
       goto refuse_caps;
     }
   } else if (!strcmp (mimetype, "audio/x-ac3")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_AC3);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_AC3);
   } else if (!strcmp (mimetype, "audio/x-eac3")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_EAC3);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_EAC3);
   } else if (!strcmp (mimetype, "audio/x-dts")) {
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_DTS);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_DTS);
   } else if (!strcmp (mimetype, "audio/x-tta")) {
     gint width;
 
@@ -1700,7 +1787,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
 
     gst_structure_get_int (structure, "width", &width);
     audiocontext->bitdepth = width;
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_TTA);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_TTA);
 
   } else if (!strcmp (mimetype, "audio/x-pn-realaudio")) {
     gint raversion;
@@ -1709,13 +1796,16 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
     gst_structure_get_int (structure, "raversion", &raversion);
     switch (raversion) {
       case 1:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4);
         break;
       case 2:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_28_8);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_AUDIO_REAL_28_8);
         break;
       case 8:
-        context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_COOK);
+        gst_matroska_mux_set_codec_id (context,
+            GST_MATROSKA_CODEC_ID_AUDIO_REAL_COOK);
         break;
       default:
         goto refuse_caps;
@@ -1816,7 +1906,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
           (guint8 *) codec_priv + WAVEFORMATEX_SIZE, -1);
     }
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_ACM);
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_ACM);
     context->codec_priv = (gpointer) codec_priv;
     context->codec_priv_size = codec_priv_size;
   }
@@ -1832,6 +1922,10 @@ refuse_caps:
   }
 }
 
+/* we probably don't have the data at start,
+ * so have to reserve (a maximum) space to write this at the end.
+ * bit spacy, but some formats can hold quite some */
+#define SUBTITLE_MAX_CODEC_PRIVATE   2048       /* must be > 128 */
 
 /**
  * gst_matroska_mux_subtitle_pad_setcaps:
@@ -1845,11 +1939,6 @@ refuse_caps:
 static gboolean
 gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
 {
-  /* FIXME:
-   * Consider this as boilerplate code for now. There is
-   * no single subtitle creation element in GStreamer,
-   * neither do I know how subtitling works at all. */
-
   /* There is now (at least) one such alement (kateenc), and I'm going
      to handle it here and claim it works when it can be piped back
      through GStreamer and VLC */
@@ -1860,6 +1949,10 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
   GstMatroskaPad *collect_pad;
   const gchar *mimetype;
   GstStructure *structure;
+  const GValue *value = NULL;
+  GstBuffer *buf = NULL;
+  gchar *id = NULL;
+  gboolean ret = TRUE;
 
   mux = GST_MATROSKA_MUX (GST_PAD_PARENT (pad));
 
@@ -1874,17 +1967,19 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
   structure = gst_caps_get_structure (caps, 0);
   mimetype = gst_structure_get_name (structure);
 
+  /* keep track of default set in request_pad */
+  id = context->codec_id;
+
   /* general setup */
   scontext->check_utf8 = 1;
   scontext->invalid_utf8 = 0;
   context->default_duration = 0;
 
-  /* TODO: - other format than Kate */
-
   if (!strcmp (mimetype, "subtitle/x-kate")) {
     const GValue *streamheader;
 
-    context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_KATE);
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_SUBTITLE_KATE);
 
     if (context->codec_priv != NULL) {
       g_free (context->codec_priv);
@@ -1896,12 +1991,64 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
     if (!kate_streamheader_to_codecdata (streamheader, context)) {
       GST_ELEMENT_ERROR (mux, STREAM, MUX, (NULL),
           ("kate stream headers missing or malformed"));
-      return FALSE;
-    }
-    return TRUE;
+      ret = FALSE;
+      goto exit;
+    }
+  } else if (!strcmp (mimetype, "text/plain")) {
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_SUBTITLE_UTF8);
+  } else if (!strcmp (mimetype, "application/x-ssa")) {
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_SSA);
+  } else if (!strcmp (mimetype, "application/x-ass")) {
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_ASS);
+  } else if (!strcmp (mimetype, "application/x-usf")) {
+    gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_USF);
+  } else if (!strcmp (mimetype, "video/x-dvd-subpicture")) {
+    gst_matroska_mux_set_codec_id (context,
+        GST_MATROSKA_CODEC_ID_SUBTITLE_VOBSUB);
+  } else {
+    id = NULL;
+    ret = FALSE;
+    goto exit;
   }
 
-  return FALSE;
+  /* maybe some private data, e.g. vobsub */
+  value = gst_structure_get_value (structure, "codec_data");
+  if (value)
+    buf = gst_value_get_buffer (value);
+  if (buf != NULL) {
+    guint8 *priv_data = NULL, *priv_buffer_data;
+    gsize priv_data_size = 0;
+
+    priv_buffer_data =
+        gst_buffer_map (buf, &priv_data_size, NULL, GST_MAP_READ);
+    if (priv_data_size > SUBTITLE_MAX_CODEC_PRIVATE) {
+      GST_WARNING_OBJECT (mux, "pad %" GST_PTR_FORMAT " subtitle private data"
+          " exceeded maximum (%d); discarding", pad,
+          SUBTITLE_MAX_CODEC_PRIVATE);
+      gst_buffer_unmap (buf, priv_data, priv_data_size);
+      return TRUE;
+    }
+
+    if (context->codec_priv != NULL)
+      g_free (context->codec_priv);
+
+    priv_data = g_malloc0 (priv_data_size);
+    memcpy (priv_data, priv_buffer_data, priv_data_size);
+    context->codec_priv = priv_data;
+    context->codec_priv_size = priv_data_size;
+    gst_buffer_unmap (buf, priv_buffer_data, priv_data_size);
+  }
+
+  GST_DEBUG_OBJECT (pad, "codec_id %s, codec data size %u",
+      GST_STR_NULL (context->codec_id), context->codec_priv_size);
+
+exit:
+  /* free default if modified */
+  if (id)
+    g_free (id);
+
+  return ret;
 }
 
 
@@ -1928,6 +2075,8 @@ gst_matroska_mux_request_new_pad (GstElement * element,
   GstMatroskaCapsFunc capsfunc = NULL;
   GstMatroskaTrackContext *context = NULL;
   gint pad_id;
+  gboolean locked = TRUE;
+  gchar *id = NULL;
 
   if (templ == gst_element_class_get_pad_template (klass, "audio_%u")) {
     /* don't mix named and unnamed pads, if the pad already exists we fail when
@@ -1971,6 +2120,9 @@ gst_matroska_mux_request_new_pad (GstElement * element,
         g_new0 (GstMatroskaTrackSubtitleContext, 1);
     context->type = GST_MATROSKA_TRACK_TYPE_SUBTITLE;
     context->name = g_strdup ("Subtitle");
+    /* setcaps may only provide proper one a lot later */
+    id = g_strdup ("S_SUB_UNKNOWN");
+    locked = FALSE;
   } else {
     GST_WARNING_OBJECT (mux, "This is not our template!");
     return NULL;
@@ -1982,24 +2134,13 @@ gst_matroska_mux_request_new_pad (GstElement * element,
 
   gst_matroskamux_pad_init (newpad);
   collect_pad = (GstMatroskaPad *)
-      gst_collect_pads_add_pad (mux->collect, GST_PAD (newpad),
-      sizeof (GstMatroskaPad),
-      (GstCollectDataDestroyNotify) gst_matroska_pad_free);
+      gst_collect_pads2_add_pad_full (mux->collect, GST_PAD (newpad),
+      sizeof (GstMatroskamuxPad),
+      (GstCollectData2DestroyNotify) gst_matroska_pad_free, locked);
 
   collect_pad->track = context;
   gst_matroska_pad_reset (collect_pad, FALSE);
-
-  /* FIXME: hacked way to override/extend the event function of
-   * GstCollectPads; because it sets its own event function giving the
-   * element no access to events.
-   * TODO GstCollectPads should really give its 'users' a clean chance to
-   * properly handle events that are not meant for collectpads itself.
-   * Perhaps a callback or so, though rejected (?) in #340060.
-   * This would allow (clean) transcoding of info from demuxer/streams
-   * to another muxer */
-  mux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
-  gst_pad_set_event_function (GST_PAD (newpad),
-      GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_sink_event));
+  collect_pad->track->codec_id = id;
 
   collect_pad->capsfunc = capsfunc;
   gst_pad_set_active (GST_PAD (newpad), TRUE);
@@ -2037,7 +2178,7 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad)
   mux = GST_MATROSKA_MUX (GST_PAD_PARENT (pad));
 
   for (walk = mux->collect->data; walk; walk = g_slist_next (walk)) {
-    GstCollectData *cdata = (GstCollectData *) walk->data;
+    GstCollectData2 *cdata = (GstCollectData2 *) walk->data;
     GstMatroskaPad *collect_pad = (GstMatroskaPad *) cdata;
 
     if (cdata->pad == pad) {
@@ -2058,7 +2199,7 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad)
     }
   }
 
-  gst_collect_pads_remove_pad (mux->collect, pad);
+  gst_collect_pads2_remove_pad (mux->collect, pad);
   if (gst_element_remove_pad (element, pad))
     mux->num_streams--;
 }
@@ -2095,6 +2236,14 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux,
         context->language);
   }
 
+  /* FIXME: until we have a nice way of getting the codecname
+   * out of the caps, I'm not going to enable this. Too much
+   * (useless, double, boring) work... */
+  /* TODO: Use value from tags if any */
+  /*gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_CODECNAME,
+     context->codec_name); */
+  gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_TRACKNAME, context->name);
+
   /* type-specific stuff */
   switch (context->type) {
     case GST_MATROSKA_TRACK_TYPE_VIDEO:{
@@ -2145,6 +2294,24 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux,
       break;
     }
 
+      /* this is what we write for now and must be filled
+       * and remainder void'ed later on */
+#define SUBTITLE_DUMMY_SIZE   (1 + 1 + 14 + 1 + 2 + SUBTITLE_MAX_CODEC_PRIVATE)
+
+    case GST_MATROSKA_TRACK_TYPE_SUBTITLE:{
+      gpointer buf;
+
+      context->pos = ebml->pos;
+      /* CodecID is mandatory ... */
+      gst_ebml_write_ascii (ebml, GST_MATROSKA_ID_CODECID, "S_SUB_UNKNOWN");
+      /* reserve space */
+      buf = g_malloc0 (SUBTITLE_MAX_CODEC_PRIVATE);
+      gst_ebml_write_binary (ebml, GST_EBML_ID_VOID, buf,
+          SUBTITLE_MAX_CODEC_PRIVATE);
+      g_free (buf);
+      /* real data has to be written at finish */
+      return;
+    }
     default:
       /* doesn't need type-specific data */
       break;
@@ -2154,13 +2321,6 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux,
   if (context->codec_priv)
     gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECPRIVATE,
         context->codec_priv, context->codec_priv_size);
-  /* FIXME: until we have a nice way of getting the codecname
-   * out of the caps, I'm not going to enable this. Too much
-   * (useless, double, boring) work... */
-  /* TODO: Use value from tags if any */
-  /*gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_CODECNAME,
-     context->codec_name); */
-  gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_TRACKNAME, context->name);
 }
 
 
@@ -2230,7 +2390,7 @@ gst_matroska_mux_start (GstMatroskaMux * mux)
     if (tags != NULL && !gst_tag_list_is_empty (tags)) {
       guint64 master_tags, master_tag;
 
-      GST_DEBUG ("Writing tags");
+      GST_DEBUG_OBJECT (mux, "Writing tags");
 
       /* TODO: maybe limit via the TARGETS id by looking at the source pad */
       mux->tags_pos = ebml->pos;
@@ -2304,7 +2464,7 @@ gst_matroska_mux_start (GstMatroskaMux * mux)
       child = gst_ebml_write_master_start (ebml, GST_MATROSKA_ID_TRACKENTRY);
       gst_matroska_mux_track_header (mux, collect_pad->track);
       gst_ebml_write_master_finish (ebml, child);
-      /* some remaing pad/track setup */
+      /* some remaining pad/track setup */
       collect_pad->default_duration_scaled =
           gst_util_uint64_scale (collect_pad->track->default_duration,
           1, mux->time_scale);
@@ -2431,7 +2591,7 @@ gst_matroska_mux_finish (GstMatroskaMux * mux)
   if (tags != NULL && !gst_tag_list_is_empty (tags)) {
     guint64 master_tags, master_tag;
 
-    GST_DEBUG ("Writing tags");
+    GST_DEBUG_OBJECT (mux, "Writing tags");
 
     /* TODO: maybe limit via the TARGETS id by looking at the source pad */
     mux->tags_pos = ebml->pos;
@@ -2479,16 +2639,23 @@ gst_matroska_mux_finish (GstMatroskaMux * mux)
     gst_ebml_write_seek (ebml, my_pos);
   }
 
-  /* update duration */
-  /* first get the overall duration */
-  /* a released track may have left a duration in here */
+  /* loop tracks:
+   * - first get the overall duration
+   *   (a released track may have left a duration in here)
+   * - write some track header data for subtitles
+   */
   duration = mux->duration;
+  pos = ebml->pos;
   for (collected = mux->collect->data; collected;
       collected = g_slist_next (collected)) {
     GstMatroskaPad *collect_pad;
     GstClockTime min_duration;  /* observed minimum duration */
+    GstMatroskaTrackContext *context;
+    gint voidleft = 0, fill = 0;
+    gpointer codec_id;
 
     collect_pad = (GstMatroskaPad *) collected->data;
+    context = collect_pad->track;
 
     GST_DEBUG_OBJECT (mux,
         "Pad %" GST_PTR_FORMAT " start ts %" GST_TIME_FORMAT
@@ -2510,7 +2677,41 @@ gst_matroska_mux_finish (GstMatroskaMux * mux)
     if (GST_CLOCK_TIME_IS_VALID (collect_pad->duration) &&
         duration < collect_pad->duration)
       duration = collect_pad->duration;
-  }
+
+    if (context->type != GST_MATROSKA_TRACK_TYPE_SUBTITLE || !context->pos)
+      continue;
+
+  again:
+    /* write subtitle type and possible private data */
+    gst_ebml_write_seek (ebml, context->pos);
+    /* complex way to write ascii to account for extra filling */
+    codec_id = g_malloc0 (strlen (context->codec_id) + 1 + fill);
+    strcpy (codec_id, context->codec_id);
+    gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECID,
+        codec_id, strlen (context->codec_id) + 1 + fill);
+    g_free (codec_id);
+    if (context->codec_priv)
+      gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECPRIVATE,
+          context->codec_priv, context->codec_priv_size);
+    voidleft = SUBTITLE_DUMMY_SIZE - (ebml->pos - context->pos);
+    /* void'ify; sigh, variable sized length field */
+    if (voidleft == 1) {
+      fill = 1;
+      goto again;
+    } else if (voidleft && voidleft <= 128)
+      gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, voidleft - 2);
+    else if (voidleft >= 130)
+      gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, voidleft - 3);
+    else if (voidleft == 129) {
+      gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, 64);
+      gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, 63);
+    }
+  }
+
+  /* seek back (optional, but do anyway) */
+  gst_ebml_write_seek (ebml, pos);
+
+  /* update duration */
   if (duration != 0) {
     GST_DEBUG_OBJECT (mux, "final total duration: %" GST_TIME_FORMAT,
         GST_TIME_ARGS (duration));
@@ -2533,77 +2734,6 @@ gst_matroska_mux_finish (GstMatroskaMux * mux)
   gst_ebml_write_master_finish (ebml, mux->segment_pos);
 }
 
-
-/**
- * gst_matroska_mux_best_pad:
- * @mux: #GstMatroskaMux
- * @popped: True if at least one buffer was popped from #GstCollectPads
- *
- * Find a pad with the oldest data
- * (data from this pad should be written first).
- *
- * Returns: Selected pad.
- */
-static GstMatroskaPad *
-gst_matroska_mux_best_pad (GstMatroskaMux * mux, gboolean * popped)
-{
-  GSList *collected;
-  GstMatroskaPad *best = NULL;
-
-  *popped = FALSE;
-  for (collected = mux->collect->data; collected;
-      collected = g_slist_next (collected)) {
-    GstMatroskaPad *collect_pad;
-
-    collect_pad = (GstMatroskaPad *) collected->data;
-    /* fetch a new buffer if needed */
-    if (collect_pad->buffer == NULL) {
-      collect_pad->buffer = gst_collect_pads_pop (mux->collect,
-          (GstCollectData *) collect_pad);
-
-      if (collect_pad->buffer != NULL) {
-        GstClockTime time;
-
-        *popped = TRUE;
-        /* convert to running time */
-        time = GST_BUFFER_TIMESTAMP (collect_pad->buffer);
-        /* invalid should pass */
-        if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (time))) {
-          time = gst_segment_to_running_time (&collect_pad->collect.segment,
-              GST_FORMAT_TIME, time);
-          if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) {
-            GST_DEBUG_OBJECT (mux, "clipping buffer on pad %s outside segment",
-                GST_PAD_NAME (collect_pad->collect.pad));
-            gst_buffer_unref (collect_pad->buffer);
-            collect_pad->buffer = NULL;
-            return NULL;
-          } else {
-            GST_LOG_OBJECT (mux, "buffer ts %" GST_TIME_FORMAT " -> %"
-                GST_TIME_FORMAT " running time",
-                GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (collect_pad->buffer)),
-                GST_TIME_ARGS (time));
-            collect_pad->buffer =
-                gst_buffer_make_writable (collect_pad->buffer);
-            GST_BUFFER_TIMESTAMP (collect_pad->buffer) = time;
-          }
-        }
-      }
-    }
-
-    /* if we have a buffer check if it is better then the current best one */
-    if (collect_pad->buffer != NULL) {
-      if (best == NULL || !GST_BUFFER_TIMESTAMP_IS_VALID (collect_pad->buffer)
-          || (GST_BUFFER_TIMESTAMP_IS_VALID (best->buffer)
-              && GST_BUFFER_TIMESTAMP (collect_pad->buffer) <
-              GST_BUFFER_TIMESTAMP (best->buffer))) {
-        best = collect_pad;
-      }
-    }
-  }
-
-  return best;
-}
-
 /**
  * gst_matroska_mux_buffer_header:
  * @track: Track context.
@@ -2749,10 +2879,11 @@ gst_matroska_mux_stop_streamheader (GstMatroskaMux * mux)
  * Returns: Result of the gst_pad_push issued to write the data.
  */
 static GstFlowReturn
-gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad)
+gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad,
+    GstBuffer * buf)
 {
   GstEbmlWrite *ebml = mux->ebml_write;
-  GstBuffer *buf, *hdr;
+  GstBuffer *hdr;
   guint64 blockgroup;
   gboolean write_duration;
   gint16 relative_timestamp;
@@ -2762,8 +2893,6 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad)
   GstMatroskamuxPad *pad;
 
   /* write data */
-  buf = collect_pad->buffer;
-  collect_pad->buffer = NULL;
   pad = GST_MATROSKAMUX_PAD_CAST (collect_pad->collect.pad);
 
   /* vorbis/theora headers are retrieved from caps and put in CodecPrivate */
@@ -2953,10 +3082,9 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad)
   }
 }
 
-
 /**
- * gst_matroska_mux_collected:
- * @pads: #GstCollectPads
+ * gst_matroska_mux_handle_buffer:
+ * @pads: #GstCollectPads2
  * @uuser_data: #GstMatroskaMux
  *
  * Collectpads callback.
@@ -2964,12 +3092,12 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad)
  * Returns: #GstFlowReturn
  */
 static GstFlowReturn
-gst_matroska_mux_collected (GstCollectPads * pads, gpointer user_data)
+gst_matroska_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * data,
+    GstBuffer * buf, gpointer user_data)
 {
   GstMatroskaMux *mux = GST_MATROSKA_MUX (user_data);
   GstEbmlWrite *ebml = mux->ebml_write;
   GstMatroskaPad *best;
-  gboolean popped;
   GstFlowReturn ret = GST_FLOW_OK;
 
   GST_DEBUG_OBJECT (mux, "Collected pads");
@@ -2988,53 +3116,53 @@ gst_matroska_mux_collected (GstCollectPads * pads, gpointer user_data)
     mux->state = GST_MATROSKA_MUX_STATE_DATA;
   }
 
-  do {
-    /* which stream to write from? */
-    best = gst_matroska_mux_best_pad (mux, &popped);
+  /* provided with stream to write from */
+  best = (GstMatroskaPad *) data;
 
-    /* if there is no best pad, we have reached EOS */
-    if (best == NULL) {
-      /* buffer popped, but none returned means it was clipped */
-      if (popped)
-        break;
-      GST_DEBUG_OBJECT (mux, "No best pad finishing...");
-      if (!mux->streamable) {
-        gst_matroska_mux_finish (mux);
-      } else {
-        GST_DEBUG_OBJECT (mux, "... but streamable, nothing to finish");
-      }
-      gst_pad_push_event (mux->srcpad, gst_event_new_eos ());
-      ret = GST_FLOW_EOS;
-      break;
+  /* if there is no best pad, we have reached EOS */
+  if (best == NULL) {
+    GST_DEBUG_OBJECT (mux, "No best pad finishing...");
+    if (!mux->streamable) {
+      gst_matroska_mux_finish (mux);
+    } else {
+      GST_DEBUG_OBJECT (mux, "... but streamable, nothing to finish");
     }
-    GST_DEBUG_OBJECT (best->collect.pad, "best pad - buffer ts %"
-        GST_TIME_FORMAT " dur %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (best->buffer)),
-        GST_TIME_ARGS (GST_BUFFER_DURATION (best->buffer)));
+    gst_pad_push_event (mux->srcpad, gst_event_new_eos ());
+    ret = GST_FLOW_EOS;
+    goto exit;
+  }
 
-    /* make note of first and last encountered timestamps, so we can calculate
-     * the actual duration later when we send an updated header on eos */
-    if (GST_BUFFER_TIMESTAMP_IS_VALID (best->buffer)) {
-      GstClockTime start_ts = GST_BUFFER_TIMESTAMP (best->buffer);
-      GstClockTime end_ts = start_ts;
+  /* if we have a best stream, should also have a buffer */
+  g_assert (buf);
 
-      if (GST_BUFFER_DURATION_IS_VALID (best->buffer))
-        end_ts += GST_BUFFER_DURATION (best->buffer);
-      else if (best->track->default_duration)
-        end_ts += best->track->default_duration;
+  GST_DEBUG_OBJECT (best->collect.pad, "best pad - buffer ts %"
+      GST_TIME_FORMAT " dur %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
 
-      if (!GST_CLOCK_TIME_IS_VALID (best->end_ts) || end_ts > best->end_ts)
-        best->end_ts = end_ts;
+  /* make note of first and last encountered timestamps, so we can calculate
+   * the actual duration later when we send an updated header on eos */
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+    GstClockTime start_ts = GST_BUFFER_TIMESTAMP (buf);
+    GstClockTime end_ts = start_ts;
 
-      if (G_UNLIKELY (best->start_ts == GST_CLOCK_TIME_NONE ||
-              start_ts < best->start_ts))
-        best->start_ts = start_ts;
-    }
+    if (GST_BUFFER_DURATION_IS_VALID (buf))
+      end_ts += GST_BUFFER_DURATION (buf);
+    else if (best->track->default_duration)
+      end_ts += best->track->default_duration;
+
+    if (!GST_CLOCK_TIME_IS_VALID (best->end_ts) || end_ts > best->end_ts)
+      best->end_ts = end_ts;
+
+    if (G_UNLIKELY (best->start_ts == GST_CLOCK_TIME_NONE ||
+            start_ts < best->start_ts))
+      best->start_ts = start_ts;
+  }
 
-    /* write one buffer */
-    ret = gst_matroska_mux_write_data (mux, best);
-  } while (ret == GST_FLOW_OK && !popped);
+  /* write one buffer */
+  ret = gst_matroska_mux_write_data (mux, best, buf);
 
+exit:
   return ret;
 }
 
@@ -3058,12 +3186,12 @@ gst_matroska_mux_change_state (GstElement * element, GstStateChange transition)
     case GST_STATE_CHANGE_NULL_TO_READY:
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_collect_pads_start (mux->collect);
+      gst_collect_pads2_start (mux->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_collect_pads_stop (mux->collect);
+      gst_collect_pads2_stop (mux->collect);
       break;
     default:
       break;
diff --git a/gst/matroska/matroska-mux.h b/gst/matroska/matroska-mux.h
index 4377f26b7..da9a3e7bf 100644
--- a/gst/matroska/matroska-mux.h
+++ b/gst/matroska/matroska-mux.h
@@ -24,7 +24,7 @@
 #define __GST_MATROSKA_MUX_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 #include "ebml-write.h"
 #include "matroska-ids.h"
@@ -58,12 +58,10 @@ typedef gboolean (*GstMatroskaCapsFunc) (GstPad *pad, GstCaps *caps);
 /* all information needed for one matroska stream */
 typedef struct
 {
-  GstCollectData collect;       /* we extend the CollectData */
+  GstCollectData2 collect;       /* we extend the CollectData */
   GstMatroskaCapsFunc capsfunc;
   GstMatroskaTrackContext *track;
 
-  GstBuffer *buffer;            /* the queued buffer for this pad */
-
   guint64 duration;
   GstClockTime start_ts;
   GstClockTime end_ts;    /* last timestamp + (if available) duration */
@@ -79,8 +77,7 @@ typedef struct _GstMatroskaMux {
 
   /* pads */
   GstPad        *srcpad;
-  GstCollectPads *collect;
-  GstPadEventFunction collect_event;
+  GstCollectPads2 *collect;
   GstEbmlWrite *ebml_write;
 
   guint          num_streams,
diff --git a/gst/matroska/matroska-read-common.c b/gst/matroska/matroska-read-common.c
index fe07e295a..0b45855df 100644
--- a/gst/matroska/matroska-read-common.c
+++ b/gst/matroska/matroska-read-common.c
@@ -999,6 +999,21 @@ gst_matroska_read_common_parse_index_cuetrack (GstMatroskaReadCommon * common,
 
   DEBUG_ELEMENT_STOP (common, ebml, "CueTrackPositions", ret);
 
+  /* (e.g.) lavf typically creates entries without a block number,
+   * which is bogus and leads to contradictory information */
+  if (common->index->len) {
+    GstMatroskaIndex *last_idx;
+
+    last_idx = &g_array_index (common->index, GstMatroskaIndex,
+        common->index->len - 1);
+    if (last_idx->block == idx.block && last_idx->pos == idx.pos &&
+        last_idx->track == idx.track && idx.time > last_idx->time) {
+      GST_DEBUG_OBJECT (common, "Cue entry refers to same location, "
+          "but has different time than previous entry; discarding");
+      idx.track = 0;
+    }
+  }
+
   if ((ret == GST_FLOW_OK || ret == GST_FLOW_EOS)
       && idx.pos != (guint64) - 1 && idx.track > 0) {
     g_array_append_val (common->index, idx);
diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c
index 50d1f976d..af192900f 100644
--- a/gst/multifile/gstmultifilesink.c
+++ b/gst/multifile/gstmultifilesink.c
@@ -655,12 +655,17 @@ write_error:
     return GST_FLOW_ERROR;
   }
 stdio_write_error:
-  {
-    GST_ELEMENT_ERROR (multifilesink, RESOURCE, WRITE,
-        ("Error while writing to file."), (NULL));
-    gst_buffer_unmap (buffer, data, size);
-    return GST_FLOW_ERROR;
+  switch (errno) {
+    case ENOSPC:
+      GST_ELEMENT_ERROR (multifilesink, RESOURCE, NO_SPACE_LEFT,
+          ("Error while writing to file."), ("%s", g_strerror (errno)));
+      break;
+    default:
+      GST_ELEMENT_ERROR (multifilesink, RESOURCE, WRITE,
+          ("Error while writing to file."), ("%s", g_strerror (errno)));
   }
+  gst_buffer_unmap (buffer, data, size);
+  return GST_FLOW_ERROR;
 }
 
 static gboolean
diff --git a/gst/multipart/multipartmux.c b/gst/multipart/multipartmux.c
index 73ddbbefd..475630485 100644
--- a/gst/multipart/multipartmux.c
+++ b/gst/multipart/multipartmux.c
@@ -84,7 +84,7 @@ static GstPad *gst_multipart_mux_request_new_pad (GstElement * element,
 static GstStateChangeReturn gst_multipart_mux_change_state (GstElement *
     element, GstStateChange transition);
 
-static GstFlowReturn gst_multipart_mux_collected (GstCollectPads * pads,
+static GstFlowReturn gst_multipart_mux_collected (GstCollectPads2 * pads,
     GstMultipartMux * mux);
 
 static void gst_multipart_mux_set_property (GObject * object, guint prop_id,
@@ -148,9 +148,9 @@ gst_multipart_mux_init (GstMultipartMux * multipart_mux)
 
   multipart_mux->boundary = g_strdup (DEFAULT_BOUNDARY);
 
-  multipart_mux->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (multipart_mux->collect,
-      (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_multipart_mux_collected),
+  multipart_mux->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_function (multipart_mux->collect,
+      (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_multipart_mux_collected),
       multipart_mux);
 }
 
@@ -194,8 +194,8 @@ gst_multipart_mux_request_new_pad (GstElement * element,
     GstMultipartPadData *multipartpad;
 
     multipartpad = (GstMultipartPadData *)
-        gst_collect_pads_add_pad (multipart_mux->collect, newpad,
-        sizeof (GstMultipartPadData), NULL);
+        gst_collect_pads2_add_pad (multipart_mux->collect, newpad,
+        sizeof (GstMultipartPadData));
 
     /* save a pointer to our data in the pad */
     multipartpad->pad = newpad;
@@ -340,7 +340,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux)
   /* try to make sure we have a buffer from each usable pad first */
   walk = mux->collect->data;
   while (walk) {
-    GstCollectData *data = (GstCollectData *) walk->data;
+    GstCollectData2 *data = (GstCollectData2 *) walk->data;
     GstMultipartPadData *pad = (GstMultipartPadData *) data;
 
     walk = g_slist_next (walk);
@@ -349,7 +349,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux)
     if (pad->buffer == NULL) {
       GstBuffer *buf = NULL;
 
-      buf = gst_collect_pads_pop (mux->collect, data);
+      buf = gst_collect_pads2_pop (mux->collect, data);
 
       /* Store timestamp with segment_start and preroll */
       if (buf && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
@@ -383,7 +383,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux)
  * 3) push both buffers on best pad, go to 1
  */
 static GstFlowReturn
-gst_multipart_mux_collected (GstCollectPads * pads, GstMultipartMux * mux)
+gst_multipart_mux_collected (GstCollectPads2 * pads, GstMultipartMux * mux)
 {
   GstMultipartPadData *best;
   GstFlowReturn ret = GST_FLOW_OK;
@@ -605,11 +605,11 @@ gst_multipart_mux_change_state (GstElement * element, GstStateChange transition)
       multipart_mux->negotiated = FALSE;
       multipart_mux->need_segment = TRUE;
       GST_DEBUG_OBJECT (multipart_mux, "starting collect pads");
-      gst_collect_pads_start (multipart_mux->collect);
+      gst_collect_pads2_start (multipart_mux->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       GST_DEBUG_OBJECT (multipart_mux, "stopping collect pads");
-      gst_collect_pads_stop (multipart_mux->collect);
+      gst_collect_pads2_stop (multipart_mux->collect);
       break;
     default:
       break;
diff --git a/gst/multipart/multipartmux.h b/gst/multipart/multipartmux.h
index d2711ac0e..4583b5098 100644
--- a/gst/multipart/multipartmux.h
+++ b/gst/multipart/multipartmux.h
@@ -23,7 +23,7 @@
 #define __GST_MULTIPART_MUX__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 #include <string.h>
 
@@ -42,7 +42,7 @@ typedef struct _GstMultipartMuxClass GstMultipartMuxClass;
 /* all information needed for one multipart stream */
 typedef struct
 {
-  GstCollectData collect;       /* we extend the CollectData */
+  GstCollectData2 collect;       /* we extend the CollectData2 */
 
   GstBuffer *buffer;            /* the queued buffer for this pad */
   GstClockTime timestamp;       /* its timestamp, converted to running_time so that we can
@@ -64,7 +64,7 @@ struct _GstMultipartMux
   GstPad *srcpad;
 
   /* sinkpads */
-  GstCollectPads *collect;
+  GstCollectPads2 *collect;
 
   gint numpads;
 
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 2575b224d..615fa3c41 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -129,6 +129,8 @@
 #include "gstrtpsession.h"
 #include "gstrtpjitterbuffer.h"
 
+#include <gst/glib-compat-private.h>
+
 GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
 #define GST_CAT_DEFAULT gst_rtp_bin_debug
 
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 0472f2c8e..e6ef09447 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -67,6 +67,8 @@
 #include "rtpjitterbuffer.h"
 #include "rtpstats.h"
 
+#include <gst/glib-compat-private.h>
+
 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
 
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index bb330ddd9..a5df3273b 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -114,6 +114,8 @@
 
 #include <gst/rtp/gstrtpbuffer.h>
 
+#include <gst/glib-compat-private.h>
+
 #include "gstrtpbin-marshal.h"
 #include "gstrtpsession.h"
 #include "rtpsession.h"
@@ -908,8 +910,13 @@ start_rtcp_thread (GstRtpSession * rtpsession)
       g_thread_join (rtpsession->priv->thread);
     /* only create a new thread if the old one was stopped. Otherwise we can
      * just reuse the currently running one. */
+#if !GLIB_CHECK_VERSION (2, 31, 0)
     rtpsession->priv->thread =
         g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error);
+#else
+    rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp-thread",
+        (GThreadFunc) rtcp_thread, rtpsession, &error);
+#endif
     rtpsession->priv->thread_stopped = FALSE;
   }
   GST_RTP_SESSION_UNLOCK (rtpsession);
diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c
index 2737c6d89..8a350ba65 100644
--- a/gst/rtpmanager/gstrtpssrcdemux.c
+++ b/gst/rtpmanager/gstrtpssrcdemux.c
@@ -44,6 +44,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <gst/rtp/gstrtpbuffer.h>
 #include <gst/rtp/gstrtcpbuffer.h>
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index a73654657..aca25fe7a 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -22,6 +22,8 @@
 #include <gst/rtp/gstrtpbuffer.h>
 #include <gst/rtp/gstrtcpbuffer.h>
 
+#include <gst/glib-compat-private.h>
+
 #include "gstrtpbin-marshal.h"
 #include "rtpsession.h"
 
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 2694fc665..bbdbf7d44 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -81,6 +81,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_UNISTD_H
 #include <unistd.h>
 #endif /* HAVE_UNISTD_H */
diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c
index a830e6a52..beb979fc6 100644
--- a/gst/smpte/gstsmpte.c
+++ b/gst/smpte/gstsmpte.c
@@ -152,7 +152,7 @@ static void gst_smpte_base_init (GstSMPTEClass * klass);
 static void gst_smpte_init (GstSMPTE * smpte);
 static void gst_smpte_finalize (GstSMPTE * smpte);
 
-static GstFlowReturn gst_smpte_collected (GstCollectPads * pads,
+static GstFlowReturn gst_smpte_collected (GstCollectPads2 * pads,
     GstSMPTE * smpte);
 
 static void gst_smpte_set_property (GObject * object, guint prop_id,
@@ -360,15 +360,15 @@ gst_smpte_init (GstSMPTE * smpte)
       gst_pad_new_from_static_template (&gst_smpte_src_template, "src");
   gst_element_add_pad (GST_ELEMENT (smpte), smpte->srcpad);
 
-  smpte->collect = gst_collect_pads_new ();
-  gst_collect_pads_set_function (smpte->collect,
-      (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_smpte_collected), smpte);
-  gst_collect_pads_start (smpte->collect);
+  smpte->collect = gst_collect_pads2_new ();
+  gst_collect_pads2_set_function (smpte->collect,
+      (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_smpte_collected), smpte);
+  gst_collect_pads2_start (smpte->collect);
 
-  gst_collect_pads_add_pad (smpte->collect, smpte->sinkpad1,
-      sizeof (GstCollectData), NULL);
-  gst_collect_pads_add_pad (smpte->collect, smpte->sinkpad2,
-      sizeof (GstCollectData), NULL);
+  gst_collect_pads2_add_pad (smpte->collect, smpte->sinkpad1,
+      sizeof (GstCollectData2));
+  gst_collect_pads2_add_pad (smpte->collect, smpte->sinkpad2,
+      sizeof (GstCollectData2));
 
   smpte->fps = DEFAULT_PROP_FPS;
   smpte->type = DEFAULT_PROP_TYPE;
@@ -441,7 +441,7 @@ gst_smpte_blend_i420 (guint8 * in1, guint8 * in2, guint8 * out, GstMask * mask,
 }
 
 static GstFlowReturn
-gst_smpte_collected (GstCollectPads * pads, GstSMPTE * smpte)
+gst_smpte_collected (GstCollectPads2 * pads, GstSMPTE * smpte)
 {
   GstBuffer *outbuf;
   GstClockTime ts;
@@ -458,14 +458,14 @@ gst_smpte_collected (GstCollectPads * pads, GstSMPTE * smpte)
       smpte->fps_denom, smpte->fps_num);
 
   for (collected = pads->data; collected; collected = g_slist_next (collected)) {
-    GstCollectData *data;
+    GstCollectData2 *data;
 
-    data = (GstCollectData *) collected->data;
+    data = (GstCollectData2 *) collected->data;
 
     if (data->pad == smpte->sinkpad1)
-      in1 = gst_collect_pads_pop (pads, data);
+      in1 = gst_collect_pads2_pop (pads, data);
     else if (data->pad == smpte->sinkpad2)
-      in2 = gst_collect_pads_pop (pads, data);
+      in2 = gst_collect_pads2_pop (pads, data);
   }
 
   if (in1 == NULL) {
@@ -623,11 +623,11 @@ gst_smpte_change_state (GstElement * element, GstStateChange transition)
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       gst_smpte_reset (smpte);
       GST_LOG_OBJECT (smpte, "starting collectpads");
-      gst_collect_pads_start (smpte->collect);
+      gst_collect_pads2_start (smpte->collect);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       GST_LOG_OBJECT (smpte, "stopping collectpads");
-      gst_collect_pads_stop (smpte->collect);
+      gst_collect_pads2_stop (smpte->collect);
       break;
     default:
       break;
diff --git a/gst/smpte/gstsmpte.h b/gst/smpte/gstsmpte.h
index dc6bc6d8a..ac138f4e7 100644
--- a/gst/smpte/gstsmpte.h
+++ b/gst/smpte/gstsmpte.h
@@ -22,7 +22,7 @@
 #define __GST_SMPTE_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstcollectpads2.h>
 
 G_BEGIN_DECLS
 
@@ -49,7 +49,7 @@ struct _GstSMPTE {
   GstPad        *srcpad,
                 *sinkpad1,
                 *sinkpad2;
-  GstCollectPads *collect;
+  GstCollectPads2 *collect;
 
   /* properties */
   gint           type;
diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c
index a302465bd..970b4aef0 100644
--- a/gst/udp/gstmultiudpsink.c
+++ b/gst/udp/gstmultiudpsink.c
@@ -41,6 +41,8 @@
 #include <errno.h>
 #include <string.h>
 
+#include "gst/glib-compat-private.h"
+
 GST_DEBUG_CATEGORY_STATIC (multiudpsink_debug);
 #define GST_CAT_DEFAULT (multiudpsink_debug)
 
diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c
index 9169d6b13..d9d45e1d8 100644
--- a/gst/udp/gstudpsrc.c
+++ b/gst/udp/gstudpsrc.c
@@ -471,12 +471,21 @@ retry:
               IOCTL_SOCKET (udpsrc->sock.fd, FIONREAD, &readsize)) < 0))
     goto ioctl_failed;
 
-  /* if we get here and there is nothing to read from the socket, the select got
-   * woken up by activity on the socket but it was not a read. We know someone
-   * will also do something with the socket so that we don't go into an infinite
-   * loop in the select(). */
+  /* If we get here and the readsize is zero, then either select was woken up
+   * by activity that is not a read, or a poll error occurred, or a UDP packet
+   * was received that has no data. Since we cannot identify which case it is,
+   * we handle all of them. This could possibly lead to a UDP packet getting
+   * lost, but since UDP is not reliable, we can accept this. */
   if (G_UNLIKELY (!readsize)) {
+    /* try to read a packet (and it will be ignored),
+     * in case a packet with no data arrived */
+    slen = sizeof (sa);
+    recvfrom (udpsrc->sock.fd, (char *) &slen, 0, 0, &sa.sa, &slen);
+
+    /* clear any error, in case a poll error occurred */
     clear_error (udpsrc);
+
+    /* poll again */
     goto retry;
   }
 
diff --git a/gst/videocrop/gstaspectratiocrop.c b/gst/videocrop/gstaspectratiocrop.c
index eb0ece2bf..1be97fa79 100644
--- a/gst/videocrop/gstaspectratiocrop.c
+++ b/gst/videocrop/gstaspectratiocrop.c
@@ -43,6 +43,8 @@
 
 #include "gstaspectratiocrop.h"
 
+#include "gst/glib-compat-private.h"
+
 GST_DEBUG_CATEGORY_STATIC (aspect_ratio_crop_debug);
 #define GST_CAT_DEFAULT aspect_ratio_crop_debug
 
diff --git a/gst/videomixer/videomixer2.c b/gst/videomixer/videomixer2.c
index e72031d5f..b8d419ae9 100644
--- a/gst/videomixer/videomixer2.c
+++ b/gst/videomixer/videomixer2.c
@@ -86,6 +86,10 @@
 #include "config.h"
 #endif
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 
 #include "videomixer2.h"
diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index e494b2fd4..1306a9a26 100644
--- a/gst/wavparse/gstwavparse.c
+++ b/gst/wavparse/gstwavparse.c
@@ -49,6 +49,11 @@
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <math.h>
 
@@ -85,6 +90,19 @@ static void gst_wavparse_loop (GstPad * pad);
 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
     GstEvent * event);
 
+static void gst_wavparse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_wavparse_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+#define DEFAULT_IGNORE_LENGTH FALSE
+
+enum
+{
+  PROP_0,
+  PROP_IGNORE_LENGTH,
+};
+
 static GstStaticPadTemplate sink_template_factory =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
@@ -113,6 +131,27 @@ gst_wavparse_class_init (GstWavParseClass * klass)
 
   object_class->dispose = gst_wavparse_dispose;
 
+  object_class->set_property = gst_wavparse_set_property;
+  object_class->get_property = gst_wavparse_get_property;
+
+  /**
+   * GstWavParse:ignore-length
+   * 
+   * This selects whether the length found in a data chunk
+   * should be ignored. This may be useful for streamed audio
+   * where the length is unknown until the end of streaming,
+   * and various software/hardware just puts some random value
+   * in there and hopes it doesn't break too much.
+   *
+   * Since: 0.10.36
+   */
+  g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
+      g_param_spec_boolean ("ignore-length",
+          "Ignore length",
+          "Ignore length from the Wave header",
+          DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
+      );
+
   gstelement_class->change_state = gst_wavparse_change_state;
   gstelement_class->send_event = gst_wavparse_send_event;
 
@@ -212,37 +251,6 @@ gst_wavparse_init (GstWavParse * wavparse)
   gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
 }
 
-/* Compute (value * nom) % denom, avoiding overflow.  This can be used
- * to perform ceiling or rounding division together with
- * gst_util_uint64_scale[_int]. */
-#define uint64_scale_modulo(val, nom, denom) \
-  ((val % denom) * (nom % denom) % denom)
-
-/* Like gst_util_uint64_scale, but performs ceiling division. */
-static guint64
-uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
-{
-  guint64 result = gst_util_uint64_scale_int (val, num, denom);
-
-  if (uint64_scale_modulo (val, num, denom) == 0)
-    return result;
-  else
-    return result + 1;
-}
-
-/* Like gst_util_uint64_scale, but performs ceiling division. */
-static guint64
-uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom)
-{
-  guint64 result = gst_util_uint64_scale (val, num, denom);
-
-  if (uint64_scale_modulo (val, num, denom) == 0)
-    return result;
-  else
-    return result + 1;
-}
-
-
 /* FIXME: why is that not in use? */
 #if 0
 static void
@@ -531,7 +539,7 @@ gst_wavparse_fmt (GstWavParse * wav)
   gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
   gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
 
-  GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
+  GST_DEBUG ("frequency %u, channels %u", wav->rate, wav->channels);
 
   return TRUE;
 
@@ -572,8 +580,8 @@ gst_wavparse_other (GstWavParse * wav)
     GST_WARNING_OBJECT (wav, "could not peek head");
     return FALSE;
   }
-  GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
-      (gchar *) & tag, length);
+  GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %u", tag,
+      (const gchar *) &tag, length);
 
   switch (tag) {
     case GST_RIFF_TAG_LIST:
@@ -724,12 +732,12 @@ gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
   }
 
   if (wav->bps > 0) {
-    *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND);
+    *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
     return TRUE;
   } else if (wav->fact) {
     guint64 bps =
         gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
-    *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND);
+    *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
     return TRUE;
   }
 
@@ -1008,7 +1016,7 @@ gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
   *size = GST_READ_UINT32_LE (data + 4);
   gst_adapter_unmap (wav->adapter);
 
-  GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
+  GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
       GST_FOURCC_ARGS (*tag));
 
   return TRUE;
@@ -1038,7 +1046,7 @@ gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
    * so we throw poor man's exception, which can be caught if caller really
    * wants to handle 0 size chunk */
   if (!(*size) || (*size) >= (1 << 30)) {
-    GST_INFO ("Invalid/unexpected chunk size %d for tag %" GST_FOURCC_FORMAT,
+    GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
         *size, GST_FOURCC_ARGS (*tag));
     /* chain should give up */
     wav->abort_buffering = TRUE;
@@ -1073,12 +1081,14 @@ gst_wavparse_calculate_duration (GstWavParse * wav)
   if (wav->bps > 0) {
     GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
     wav->duration =
-        uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps);
+        gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
+        (guint64) wav->bps);
     GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
         GST_TIME_ARGS (wav->duration));
     return TRUE;
   } else if (wav->fact) {
-    wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate);
+    wav->duration =
+        gst_util_uint64_scale_int_ceil (GST_SECOND, wav->fact, wav->rate);
     GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
         GST_TIME_ARGS (wav->duration));
     return TRUE;
@@ -1211,12 +1221,12 @@ gst_wavparse_stream_headers (GstWavParse * wav)
       {
         /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
          * bitrate inside the mpeg stream */
-        GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
+        GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
         wav->bps = 0;
         break;
       }
       case GST_RIFF_WAVE_FORMAT_PCM:
-        if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0))
+        if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
           goto invalid_blockalign;
         /* fall through */
       default:
@@ -1298,7 +1308,11 @@ gst_wavparse_stream_headers (GstWavParse * wav)
      */
     switch (tag) {
       case GST_RIFF_TAG_data:{
-        GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
+        GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
+        if (wav->ignore_length) {
+          GST_DEBUG_OBJECT (wav, "Ignoring length");
+          size = 0;
+        }
         if (wav->streaming) {
           gst_adapter_flush (wav->adapter, 8);
           gotdata = TRUE;
@@ -1323,7 +1337,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
           /* We will continue parsing tags 'till end */
           wav->offset += size;
         }
-        GST_DEBUG_OBJECT (wav, "datasize = %d", size);
+        GST_DEBUG_OBJECT (wav, "datasize = %u", size);
         break;
       }
       case GST_RIFF_TAG_fact:{
@@ -1337,7 +1351,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
               /* need more data */
               goto exit;
             }
-            GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
+            GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
                 data_size, size);
             break;
           }
@@ -1385,7 +1399,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
             /* need more data */
             goto exit;
           }
-          GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
+          GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
               data_size, size);
           break;
         }
@@ -1520,18 +1534,20 @@ gst_wavparse_stream_headers (GstWavParse * wav)
     wav->bps =
         (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
         (guint64) wav->fact);
-    GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps);
+    GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
 #endif
     wav->vbr = TRUE;
   }
 
   if (gst_wavparse_calculate_duration (wav)) {
     gst_segment_init (&wav->segment, GST_FORMAT_TIME);
-    wav->segment.duration = wav->duration;
+    if (!wav->ignore_length)
+      wav->segment.duration = wav->duration;
   } else {
     /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
     gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
-    wav->segment.duration = wav->datasize;
+    if (!wav->ignore_length)
+      wav->segment.duration = wav->datasize;
   }
 
   /* now we have all the info to perform a pending seek if any, if no
@@ -1558,7 +1574,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
   if (wav->blockalign > 0)
     wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
 
-  GST_DEBUG_OBJECT (wav, "max buffer size %d", wav->max_buf_size);
+  GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
 
   return GST_FLOW_OK;
 
@@ -1607,7 +1623,7 @@ invalid_blockalign:
   {
     GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
         ("Stream claims blockalign = %u, which is more than %u - invalid data",
-            wav->blockalign, wav->channels * (guint) ceil (wav->depth / 8.0)));
+            wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
     goto fail;
   }
 invalid_bps:
@@ -1626,7 +1642,7 @@ no_bytes_per_sample:
 unknown_format:
   {
     GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
-        ("No caps found for format 0x%x, %d channels, %d Hz",
+        ("No caps found for format 0x%x, %u channels, %u Hz",
             wav->format, wav->channels, wav->rate));
     goto fail;
   }
@@ -1818,13 +1834,13 @@ iterate_adapter:
     if (G_UNLIKELY (extra)) {
       extra = wav->bytes_per_sample - extra;
       if (extra <= avail) {
-        GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra);
+        GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
         gst_adapter_flush (wav->adapter, extra);
         wav->offset += extra;
         wav->dataleft -= extra;
         goto iterate_adapter;
       } else {
-        GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail);
+        GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
         gst_adapter_clear (wav->adapter);
         wav->offset += avail;
         wav->dataleft -= avail;
@@ -1833,7 +1849,7 @@ iterate_adapter:
     }
 
     if (avail < desired) {
-      GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
+      GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
       return GST_FLOW_OK;
     }
 
@@ -1885,9 +1901,10 @@ iterate_adapter:
 
   if (wav->bps > 0) {
     /* and timestamps if we have a bitrate, be careful for overflows */
-    timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps);
+    timestamp =
+        gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
     next_timestamp =
-        uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps);
+        gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
     duration = next_timestamp - timestamp;
 
     /* update current running segment position */
@@ -1897,8 +1914,8 @@ iterate_adapter:
     guint64 bps =
         gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
     /* and timestamps if we have a bitrate, be careful for overflows */
-    timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps);
-    next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps);
+    timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
+    next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
     duration = next_timestamp - timestamp;
   } else {
     /* no bitrate, all we know is that the first sample has timestamp 0, all
@@ -2191,10 +2208,12 @@ gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
           if (bps) {
             if (start >= 0)
               start =
-                  uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps);
+                  gst_util_uint64_scale_ceil (start, GST_SECOND,
+                  (guint64) wav->bps);
             if (stop >= 0)
               stop =
-                  uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps);
+                  gst_util_uint64_scale_ceil (stop, GST_SECOND,
+                  (guint64) wav->bps);
           }
         }
       } else {
@@ -2319,13 +2338,14 @@ gst_wavparse_pad_convert (GstPad * pad,
               "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
               wavparse->offset);
           if (wavparse->bps > 0)
-            *dest_value = uint64_ceiling_scale (src_value, GST_SECOND,
+            *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
                 (guint64) wavparse->bps);
           else if (wavparse->fact) {
-            guint64 bps = uint64_ceiling_scale_int (wavparse->datasize,
+            guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
                 wavparse->rate, wavparse->fact);
 
-            *dest_value = uint64_ceiling_scale_int (src_value, GST_SECOND, bps);
+            *dest_value =
+                gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
           } else {
             res = FALSE;
           }
@@ -2438,6 +2458,11 @@ gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
       gint64 duration = 0;
       GstFormat format;
 
+      if (wav->ignore_length) {
+        res = FALSE;
+        break;
+      }
+
       gst_query_parse_duration (query, &format, NULL);
 
       switch (format) {
@@ -2626,6 +2651,43 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition)
   return ret;
 }
 
+static void
+gst_wavparse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstWavParse *self;
+
+  g_return_if_fail (GST_IS_WAVPARSE (object));
+  self = GST_WAVPARSE (object);
+
+  switch (prop_id) {
+    case PROP_IGNORE_LENGTH:
+      self->ignore_length = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
+  }
+
+}
+
+static void
+gst_wavparse_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstWavParse *self;
+
+  g_return_if_fail (GST_IS_WAVPARSE (object));
+  self = GST_WAVPARSE (object);
+
+  switch (prop_id) {
+    case PROP_IGNORE_LENGTH:
+      g_value_set_boolean (value, self->ignore_length);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
+  }
+}
+
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
diff --git a/gst/wavparse/gstwavparse.h b/gst/wavparse/gstwavparse.h
index 54d0a0fd5..82838e4cb 100644
--- a/gst/wavparse/gstwavparse.h
+++ b/gst/wavparse/gstwavparse.h
@@ -116,6 +116,8 @@ struct _GstWavParse {
   gboolean first;
   /* discont after seek */
   gboolean discont;
+
+  gboolean ignore_length;
 };
 
 struct _GstWavParseClass {
diff --git a/po/LINGUAS b/po/LINGUAS
index 44f55d97e..73d8b85b2 100644
--- a/po/LINGUAS
+++ b/po/LINGUAS
@@ -1 +1 @@
-af az bg ca cs da de el en_GB es eu fi fr gl hu id it ja lt lv mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN zh_HK zh_TW
+af az bg ca cs da de el en_GB eo es eu fi fr gl hu id it ja lt lv mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN zh_HK zh_TW
diff --git a/po/eo.po b/po/eo.po
new file mode 100644
index 000000000..73f84a393
--- /dev/null
+++ b/po/eo.po
@@ -0,0 +1,832 @@
+# Esperanto translation for gst-plugins-good.
+# Copyright (C) 2011 Free Software Foundation, Inc.
+# This file is distributed under the same license as the gst-plugins-good package.
+# Kristjan SCHMIDT <kristjan.schmidt@googlemail.com>, 2011.
+#
+msgid ""
+msgstr ""
+"Project-Id-Version: gst-plugins-good 0.10.28.2\n"
+"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
+"POT-Creation-Date: 2011-04-16 16:25+0100\n"
+"PO-Revision-Date: 2011-06-04 21:48+0100\n"
+"Last-Translator: Kristjan SCHMIDT <kristjan.schmidt@googlemail.com>\n"
+"Language-Team: Esperanto <translation-team-eo@lists.sourceforge.net>\n"
+"Language: eo\n"
+"MIME-Version: 1.0\n"
+"Content-Type: text/plain; charset=UTF-8\n"
+"Content-Transfer-Encoding: 8bit\n"
+"Plural-Forms: nplurals=2; plural=(n != 1);\n"
+
+#: ext/esd/esdsink.c:253 ext/esd/esdsink.c:358
+msgid "Could not establish connection to sound server"
+msgstr ""
+
+#: ext/esd/esdsink.c:260
+msgid "Failed to query sound server capabilities"
+msgstr ""
+
+#. TRANSLATORS: 'song title' by 'artist name'
+#: ext/pulse/pulsesink.c:2686
+#, c-format
+msgid "'%s' by '%s'"
+msgstr "'%s' de '%s'"
+
+#: ext/flac/gstflacdec.c:1117 ext/libpng/gstpngdec.c:349
+#: ext/libpng/gstpngdec.c:360 ext/libpng/gstpngdec.c:559
+#: ext/wavpack/gstwavpackparse.c:1170 gst/avi/gstavidemux.c:5203
+msgid "Internal data stream error."
+msgstr "Interna datumflu-eraro."
+
+#: ext/jpeg/gstjpegdec.c:286
+msgid "Failed to decode JPEG image"
+msgstr ""
+
+#: ext/shout2/gstshout2.c:578
+msgid "Could not connect to server"
+msgstr "Ne eblis konekti al servilo"
+
+#: ext/soup/gstsouphttpsrc.c:871
+msgid "Server does not support seeking."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1078
+msgid "Could not resolve server name."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1084
+msgid "Could not establish connection to server."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1089
+msgid "Secure connection setup failed."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1094
+msgid "A network error occured, or the server closed the connection unexpectedly."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1100
+msgid "Server sent bad data."
+msgstr ""
+
+#: ext/soup/gstsouphttpsrc.c:1249
+msgid "No URL set."
+msgstr ""
+
+#: gst/avi/gstavimux.c:1810
+msgid "No or invalid input audio, AVI stream will be corrupt."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:519 gst/quicktime/qtdemux.c:523
+msgid "This file contains no playable streams."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:558 gst/quicktime/qtdemux.c:4009
+#: gst/quicktime/qtdemux.c:4071 gst/quicktime/qtdemux.c:4219
+msgid "This file is invalid and cannot be played."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:2439 gst/quicktime/qtdemux.c:2515
+#: gst/quicktime/qtdemux.c:2558 gst/quicktime/qtdemux.c:4798
+#: gst/quicktime/qtdemux.c:4805 gst/quicktime/qtdemux.c:5391
+#: gst/quicktime/qtdemux.c:5817 gst/quicktime/qtdemux.c:5824
+#: gst/quicktime/qtdemux.c:7305
+msgid "This file is corrupt and cannot be played."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:2647
+msgid "Invalid atom size."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:2716
+msgid "This file is incomplete and cannot be played."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:4994
+msgid "The video in this file might not play correctly."
+msgstr ""
+
+#: gst/quicktime/qtdemux.c:7334
+#, c-format
+msgid "This file contains too many streams. Only playing first %d"
+msgstr ""
+
+#: gst/rtsp/gstrtspsrc.c:5187
+msgid "No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams."
+msgstr ""
+
+#: gst/rtsp/gstrtspsrc.c:5192
+msgid "No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin."
+msgstr ""
+
+#: gst/wavparse/gstwavparse.c:2103
+msgid "Internal data flow error."
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:98 sys/oss4/oss4-mixer.c:722
+#: sys/sunaudio/gstsunaudiomixertrack.c:69
+msgid "Volume"
+msgstr "Laŭteco"
+
+#: sys/oss/gstossmixertrack.c:99 sys/oss4/oss4-mixer.c:735
+msgid "Bass"
+msgstr "Baso"
+
+#: sys/oss/gstossmixertrack.c:100 sys/oss4/oss4-mixer.c:736
+msgid "Treble"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:101
+msgid "Synth"
+msgstr "Sintezilo"
+
+#: sys/oss/gstossmixertrack.c:102 sys/oss4/oss4-mixer.c:750
+msgid "PCM"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:103
+msgid "Speaker"
+msgstr "Parolilo"
+
+#: sys/oss/gstossmixertrack.c:104
+msgid "Line-in"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:105 sys/oss4/oss4-mixer.c:741
+msgid "Microphone"
+msgstr "Mikrofono"
+
+#: sys/oss/gstossmixertrack.c:106
+msgid "CD"
+msgstr "KD"
+
+#: sys/oss/gstossmixertrack.c:107
+msgid "Mixer"
+msgstr "Miksilo"
+
+#: sys/oss/gstossmixertrack.c:108
+msgid "PCM-2"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:109
+msgid "Record"
+msgstr "Registri"
+
+#: sys/oss/gstossmixertrack.c:110
+msgid "In-gain"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:111
+msgid "Out-gain"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:112
+msgid "Line-1"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:113
+msgid "Line-2"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:114
+msgid "Line-3"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:115
+msgid "Digital-1"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:116
+msgid "Digital-2"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:117
+msgid "Digital-3"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:118
+msgid "Phone-in"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:119
+msgid "Phone-out"
+msgstr ""
+
+#: sys/oss/gstossmixertrack.c:120
+msgid "Video"
+msgstr "Video"
+
+#: sys/oss/gstossmixertrack.c:121
+msgid "Radio"
+msgstr "Radio"
+
+#: sys/oss/gstossmixertrack.c:122 sys/oss4/oss4-mixer.c:764
+#: sys/sunaudio/gstsunaudiomixertrack.c:71
+msgid "Monitor"
+msgstr "Ekrano"
+
+#: sys/oss/gstosssink.c:399 sys/oss4/oss4-sink.c:494
+#: sys/oss4/oss4-source.c:361
+msgid "Could not open audio device for playback. Device is being used by another application."
+msgstr "Ne eblis malfermi la sonaparaton por reproduktado. Ĝi estas uzate de alia aplikaĵo."
+
+#: sys/oss/gstosssink.c:406 sys/oss4/oss4-sink.c:504
+#: sys/oss4/oss4-source.c:371
+msgid "Could not open audio device for playback. You don't have permission to open the device."
+msgstr ""
+
+#: sys/oss/gstosssink.c:414 sys/oss4/oss4-sink.c:515
+#: sys/oss4/oss4-source.c:382
+msgid "Could not open audio device for playback."
+msgstr "Ne eblis malfermi la sonaparaton por reproduktado."
+
+#: sys/oss/gstosssrc.c:370
+msgid "Could not open audio device for recording. You don't have permission to open the device."
+msgstr ""
+
+#: sys/oss/gstosssrc.c:378
+msgid "Could not open audio device for recording."
+msgstr "Ne eblis malfermi sonaparaton por registrado."
+
+#: sys/oss4/oss4-mixer.c:302
+msgid "Could not open audio device for mixer control handling."
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:316
+msgid "Could not open audio device for mixer control handling. This version of the Open Sound System is not supported by this element."
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:723
+msgid "Master"
+msgstr "Ĉefe"
+
+#: sys/oss4/oss4-mixer.c:724
+msgid "Front"
+msgstr "Antaŭe"
+
+#: sys/oss4/oss4-mixer.c:725
+msgid "Rear"
+msgstr "Malantaŭe"
+
+#: sys/oss4/oss4-mixer.c:726
+msgid "Headphones"
+msgstr "Kaptelefono"
+
+#: sys/oss4/oss4-mixer.c:727
+msgid "Center"
+msgstr "Centre"
+
+#: sys/oss4/oss4-mixer.c:728
+msgid "LFE"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:729
+msgid "Surround"
+msgstr "Ĉirkaŭe"
+
+#: sys/oss4/oss4-mixer.c:730
+msgid "Side"
+msgstr "Flanke"
+
+#: sys/oss4/oss4-mixer.c:731 sys/sunaudio/gstsunaudiomixertrack.c:72
+msgid "Built-in Speaker"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:732 sys/sunaudio/gstsunaudiomixertrack.c:76
+msgid "AUX 1 Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:733 sys/sunaudio/gstsunaudiomixertrack.c:77
+msgid "AUX 2 Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:734
+msgid "AUX Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:737
+msgid "3D Depth"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:738
+msgid "3D Center"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:739
+msgid "3D Enhance"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:740
+msgid "Telephone"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:742 sys/sunaudio/gstsunaudiomixertrack.c:74
+msgid "Line Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:743 sys/oss4/oss4-mixer.c:744
+msgid "Line In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:745
+msgid "Internal CD"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:746
+msgid "Video In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:747
+msgid "AUX 1 In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:748
+msgid "AUX 2 In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:749
+msgid "AUX In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:751 sys/oss4/oss4-mixer.c:752
+msgid "Record Gain"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:753
+msgid "Output Gain"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:754
+msgid "Microphone Boost"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:755
+msgid "Loopback"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:756
+msgid "Diagnostic"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:757
+msgid "Bass Boost"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:758
+msgid "Playback Ports"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:759
+msgid "Input"
+msgstr "Enigo"
+
+#: sys/oss4/oss4-mixer.c:760 sys/oss4/oss4-mixer.c:761
+msgid "Record Source"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:762
+msgid "Monitor Source"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:763
+msgid "Keyboard Beep"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:765
+msgid "Simulate Stereo"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:766 sys/oss4/oss4-mixer.c:786
+msgid "Stereo"
+msgstr "Dukanale"
+
+#: sys/oss4/oss4-mixer.c:767
+msgid "Surround Sound"
+msgstr "Ĉirkaŭa sono"
+
+#: sys/oss4/oss4-mixer.c:768
+msgid "Microphone Gain"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:769
+msgid "Speaker Source"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:770
+msgid "Microphone Source"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:771
+msgid "Jack"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:772
+msgid "Center / LFE"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:773
+msgid "Stereo Mix"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:774
+msgid "Mono Mix"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:775
+msgid "Input Mix"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:776
+msgid "SPDIF In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:777 sys/sunaudio/gstsunaudiomixertrack.c:75
+msgid "SPDIF Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:778
+msgid "Microphone 1"
+msgstr "Mikrofono 1"
+
+#: sys/oss4/oss4-mixer.c:779
+msgid "Microphone 2"
+msgstr "Mikrofono 2"
+
+#: sys/oss4/oss4-mixer.c:780
+msgid "Digital Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:781
+msgid "Digital In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:782
+msgid "HDMI"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:783
+msgid "Modem"
+msgstr "Modemo"
+
+#: sys/oss4/oss4-mixer.c:784
+msgid "Handset"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:785
+msgid "Other"
+msgstr "Alia"
+
+#: sys/oss4/oss4-mixer.c:787
+msgid "None"
+msgstr "Neniu"
+
+#: sys/oss4/oss4-mixer.c:788
+msgid "On"
+msgstr "Enŝaltite"
+
+#: sys/oss4/oss4-mixer.c:789
+msgid "Off"
+msgstr "Elŝaltite"
+
+#: sys/oss4/oss4-mixer.c:790
+msgid "Mute"
+msgstr "Silentigi"
+
+#: sys/oss4/oss4-mixer.c:791
+msgid "Fast"
+msgstr "Rapide"
+
+#. TRANSLATORS: "Very Low" is a quality setting here
+#: sys/oss4/oss4-mixer.c:793
+msgid "Very Low"
+msgstr "Tre malalte"
+
+#. TRANSLATORS: "Low" is a quality setting here
+#: sys/oss4/oss4-mixer.c:795
+msgid "Low"
+msgstr "Malalte"
+
+#. TRANSLATORS: "Medium" is a quality setting here
+#: sys/oss4/oss4-mixer.c:797
+msgid "Medium"
+msgstr "Meze"
+
+#. TRANSLATORS: "High" is a quality setting here
+#: sys/oss4/oss4-mixer.c:799
+msgid "High"
+msgstr "Alte"
+
+#. TRANSLATORS: "Very High" is a quality setting here
+#: sys/oss4/oss4-mixer.c:801 sys/oss4/oss4-mixer.c:802
+msgid "Very High"
+msgstr "Tre alte"
+
+#. TRANSLATORS: "Production" is a quality setting here
+#: sys/oss4/oss4-mixer.c:804
+msgid "Production"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:805
+msgid "Front Panel Microphone"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:806
+msgid "Front Panel Line In"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:807
+msgid "Front Panel Headphones"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:808
+msgid "Front Panel Line Out"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:809
+msgid "Green Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:810
+msgid "Pink Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:811
+msgid "Blue Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:812
+msgid "White Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:813
+msgid "Black Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:814
+msgid "Gray Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:815
+msgid "Orange Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:816
+msgid "Red Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:817
+msgid "Yellow Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:818
+msgid "Green Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:819
+msgid "Pink Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:820
+msgid "Blue Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:821
+msgid "White Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:822
+msgid "Black Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:823
+msgid "Gray Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:824
+msgid "Orange Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:825
+msgid "Red Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:826
+msgid "Yellow Front Panel Connector"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:827
+msgid "Spread Output"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:828
+msgid "Downmix"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:872
+msgid "Virtual Mixer Input"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:874
+msgid "Virtual Mixer Output"
+msgstr ""
+
+#: sys/oss4/oss4-mixer.c:876
+msgid "Virtual Mixer Channels"
+msgstr ""
+
+#. TRANSLATORS: name + number of a volume mixer control
+#: sys/oss4/oss4-mixer.c:927
+#, c-format
+msgid "%s %d Function"
+msgstr "%s %d funkcio"
+
+#. TRANSLATORS: name of a volume mixer control
+#: sys/oss4/oss4-mixer.c:934
+#, c-format
+msgid "%s Function"
+msgstr "%s funcio"
+
+#: sys/oss4/oss4-sink.c:524 sys/oss4/oss4-source.c:392
+msgid "Could not open audio device for playback. This version of the Open Sound System is not supported by this element."
+msgstr ""
+
+#: sys/oss4/oss4-sink.c:640
+msgid "Playback is not supported by this audio device."
+msgstr ""
+
+#: sys/oss4/oss4-sink.c:647
+msgid "Audio playback error."
+msgstr ""
+
+#: sys/oss4/oss4-source.c:514
+msgid "Recording is not supported by this audio device."
+msgstr ""
+
+#: sys/oss4/oss4-source.c:521
+msgid "Error recording from audio device."
+msgstr ""
+
+#: sys/sunaudio/gstsunaudiomixertrack.c:70
+msgid "Gain"
+msgstr ""
+
+#: sys/sunaudio/gstsunaudiomixertrack.c:73
+msgid "Headphone"
+msgstr ""
+
+#: sys/v4l2/gstv4l2src.c:887
+#, c-format
+msgid "Error reading %d bytes from device '%s'."
+msgstr ""
+
+#: sys/v4l2/gstv4l2src.c:913
+#, c-format
+msgid "Got unexpected frame size of %u instead of %u."
+msgstr ""
+
+#: sys/v4l2/gstv4l2src.c:931
+#, c-format
+msgid "Error reading %d bytes on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:99
+#, c-format
+msgid "Error getting capabilities for device '%s': It isn't a v4l2 driver. Check if it is a v4l1 driver."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:139
+#, c-format
+msgid "Failed to query attributes of input %d in device %s"
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:169
+#, c-format
+msgid "Failed to get setting of tuner %d on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:213
+#, c-format
+msgid "Failed to query norm on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:264 sys/v4l2/v4l2_calls.c:347
+#, c-format
+msgid "Failed getting controls attributes on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:492
+#, c-format
+msgid "Cannot identify device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:499
+#, c-format
+msgid "This isn't a device '%s'."
+msgstr "Tio ne estas '%s'-aparato."
+
+#: sys/v4l2/v4l2_calls.c:506
+#, c-format
+msgid "Could not open device '%s' for reading and writing."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:513
+#, c-format
+msgid "Device '%s' is not a capture device."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:522
+#, c-format
+msgid "Device '%s' is not a output device."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:622
+#, c-format
+msgid "Failed to set norm for device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:660
+#, c-format
+msgid "Failed to get current tuner frequency for device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:702
+#, c-format
+msgid "Failed to set current tuner frequency for device '%s' to %lu Hz."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:736
+#, c-format
+msgid "Failed to get signal strength for device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:772
+#, c-format
+msgid "Failed to get value for control %d on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:807
+#, c-format
+msgid "Failed to set value %d for control %d on device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:839
+#, c-format
+msgid "Failed to get current input on device '%s'. May be it is a radio device"
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:864
+#, c-format
+msgid "Failed to set input %d on device %s."
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:896
+#, c-format
+msgid "Failed to get current output on device '%s'. May be it is a radio device"
+msgstr ""
+
+#: sys/v4l2/v4l2_calls.c:921
+#, c-format
+msgid "Failed to set output %d on device %s."
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:81
+#, c-format
+msgid "Could not enqueue buffers in device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:197
+#, c-format
+msgid "Failed trying to get video frames from device '%s'."
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:199
+#, c-format
+msgid "Failed after %d tries. device %s. system error: %s"
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:239
+#, c-format
+msgid "Could not get parameters on device '%s'"
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:267
+msgid "Video input device did not accept new frame rate setting."
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:339
+#, c-format
+msgid "Could not map buffers from device '%s'"
+msgstr ""
+
+#: sys/v4l2/v4l2src_calls.c:347
+#, c-format
+msgid "The driver of device '%s' does not support any known capture method."
+msgstr ""
+
+#: sys/ximage/gstximagesrc.c:719
+msgid "Changing resolution at runtime is not yet supported."
+msgstr ""
+
+#: sys/ximage/gstximagesrc.c:733
+msgid "Cannot operate without a clock"
+msgstr ""
diff --git a/sys/Makefile.am b/sys/Makefile.am
index ab08d046d..7ccd9e77a 100644
--- a/sys/Makefile.am
+++ b/sys/Makefile.am
@@ -23,6 +23,12 @@ else
 DIRECTSOUND_DIR=
 endif
 
+if USE_WAVEFORM
+WAVEFORM_DIR=waveform
+else
+WAVEFORM_DIR=
+endif
+
 if USE_SUNAUDIO
 SUNAUDIO_DIR=sunaudio
 else
@@ -77,7 +83,7 @@ else
 XIMAGE_DIR=
 endif
 
-SUBDIRS=$(DIRECTSOUND_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(SUNAUDIO_DIR) $(V4L2_DIR) $(XIMAGE_DIR)
+SUBDIRS=$(DIRECTSOUND_DIR) $(WAVEFORM_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(SUNAUDIO_DIR) $(V4L2_DIR) $(XIMAGE_DIR)
 
 DIST_SUBDIRS=directsound oss oss4 osxaudio osxvideo sunaudio v4l2 waveform ximage
 
diff --git a/sys/oss4/oss4-mixer.c b/sys/oss4/oss4-mixer.c
index e190987d6..e384452ec 100644
--- a/sys/oss4/oss4-mixer.c
+++ b/sys/oss4/oss4-mixer.c
@@ -54,6 +54,7 @@
 
 #include <gst/interfaces/mixer.h>
 #include <gst/gst-i18n-plugin.h>
+#include "gst/glib-compat-private.h"
 
 #include <glib/gprintf.h>
 
@@ -541,8 +542,13 @@ gst_oss4_mixer_start_watch_task (GstOss4Mixer * mixer)
   mixer->watch_cond = g_cond_new ();
   mixer->watch_shutdown = FALSE;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   mixer->watch_thread = g_thread_create (gst_oss4_mixer_watch_thread,
       gst_object_ref (mixer), TRUE, &err);
+#else
+  mixer->watch_thread = g_thread_try_new ("oss4-mixer-thread",
+      gst_oss4_mixer_watch_thread, gst_object_ref (mixer), &err);
+#endif
 
   if (mixer->watch_thread == NULL) {
     GST_ERROR_OBJECT (mixer, "Could not create watch thread: %s", err->message);
diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c
index 6b9074522..d25d7ae69 100644
--- a/sys/v4l2/gstv4l2bufferpool.c
+++ b/sys/v4l2/gstv4l2bufferpool.c
@@ -40,6 +40,7 @@
 #include "gstv4l2sink.h"
 #include "v4l2_calls.h"
 #include "gst/gst-i18n-plugin.h"
+#include <gst/glib-compat-private.h>
 
 /* videodev2.h is not versioned and we can't easily check for the presence
  * of enum values at compile time, but the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define
diff --git a/sys/v4l2/gstv4l2videooverlay.c b/sys/v4l2/gstv4l2videooverlay.c
index 3f28fdcd8..6ac15cdf4 100644
--- a/sys/v4l2/gstv4l2videooverlay.c
+++ b/sys/v4l2/gstv4l2videooverlay.c
@@ -39,6 +39,7 @@
 #include "v4l2_calls.h"
 
 #include "gst/gst-i18n-plugin.h"
+#include <gst/glib-compat-private.h>
 
 struct _GstV4l2Xv
 {
diff --git a/sys/waveform/Makefile.am b/sys/waveform/Makefile.am
index a9562a818..9a9fac3d7 100644
--- a/sys/waveform/Makefile.am
+++ b/sys/waveform/Makefile.am
@@ -1,19 +1,15 @@
 plugin_LTLIBRARIES = libgstwaveformsink.la
 
-# FIXME: Replace DIRECTSOUND CFLAGS+LIBS with waveform related ones and fix
-# the configure.ac + sys/Makefile.am to get this stuff building in MingW
-# For now, it's just disted for use in the VS builds.
-
 libgstwaveformsink_la_SOURCES =  gstwaveformsink.c gstwaveformplugin.c
 libgstwaveformsink_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) \
-        $(GST_PLUGINS_BASE_CFLAGS) $(DIRECTSOUND_CFLAGS)
+        $(GST_PLUGINS_BASE_CFLAGS)
 libgstwaveformsink_la_LIBADD = \
 	$(GST_PLUGINS_BASE_LIBS) \
         -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \
         $(GST_BASE_LIBS) \
 	$(GST_LIBS) \
-	$(DIRECTSOUND_LIBS)
-libgstwaveformsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(DIRECTSOUND_LDFLAGS)
+	-lwinmm
+libgstwaveformsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstwaveformsink_la_LIBTOOLFLAGS = --tag=disable-static
 
 noinst_HEADERS = gstwaveformsink.h
diff --git a/sys/waveform/gstwaveformsink.c b/sys/waveform/gstwaveformsink.c
index 36542c182..e4b7016bb 100644
--- a/sys/waveform/gstwaveformsink.c
+++ b/sys/waveform/gstwaveformsink.c
@@ -49,10 +49,6 @@
 
 GST_DEBUG_CATEGORY_STATIC (waveformsink_debug);
 
-static void gst_waveform_sink_base_init (gpointer g_class);
-static void gst_waveform_sink_class_init (GstWaveFormSinkClass * klass);
-static void gst_waveform_sink_init (GstWaveFormSink * wfsink,
-    GstWaveFormSinkClass * g_class);
 static void gst_waveform_sink_finalise (GObject * object);
 static void gst_waveform_sink_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec);
@@ -154,7 +150,7 @@ static void
 gst_waveform_sink_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
-  GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object);
+  /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); */
 
   switch (prop_id) {
     default:
@@ -167,7 +163,7 @@ static void
 gst_waveform_sink_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
-  GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object);
+  /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); */
 
   switch (prop_id) {
     default:
@@ -348,7 +344,7 @@ gst_waveform_sink_getcaps (GstBaseSink * bsink)
 static gboolean
 gst_waveform_sink_open (GstAudioSink * asink)
 {
-  GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink);
+  /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); */
 
   /* nothing to do here as the device needs to be opened with the format we will use */
 
@@ -448,7 +444,7 @@ gst_waveform_sink_unprepare (GstAudioSink * asink)
 static gboolean
 gst_waveform_sink_close (GstAudioSink * asink)
 {
-  GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink);
+  /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); */
 
   return TRUE;
 }
diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c
index 91e3f1dbe..41559c5d5 100644
--- a/sys/ximage/gstximagesrc.c
+++ b/sys/ximage/gstximagesrc.c
@@ -49,6 +49,8 @@
 #include <gst/gst.h>
 #include <gst/gst-i18n-plugin.h>
 
+#include "gst/glib-compat-private.h"
+
 GST_DEBUG_CATEGORY_STATIC (gst_debug_ximage_src);
 #define GST_CAT_DEFAULT gst_debug_ximage_src
 
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 426e6196b..ec2bbb499 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -147,6 +147,7 @@ check_PROGRAMS = \
 	elements/shapewipe \
 	elements/spectrum \
 	elements/udpsink \
+	elements/udpsrc \
 	elements/videocrop \
 	elements/videofilter \
 	elements/y4menc \
@@ -296,6 +297,9 @@ elements_sunaudio_LDADD = \
 	$(GST_PLUGINS_BASE_LIBS) -lgstinterfaces-@GST_MAJORMINOR@ \
 	$(LDADD)
 
+elements_udpsrc_CFLAGS = $(AM_CFLAGS) $(GIO_CFLAGS)
+elements_udpsrc_LDADD = $(LDADD) $(GIO_LIBS)
+
 elements_videocrop_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) $(LDADD)
 elements_videocrop_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index c868fc81e..1e2c8b2e4 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -53,6 +53,7 @@ souphttpsrc
 spectrum
 sunaudio
 udpsink
+udpsrc
 videocrop
 videofilter
 wavpackdec
diff --git a/tests/check/elements/qtmux.c b/tests/check/elements/qtmux.c
index 5ceb3d15b..84614a265 100644
--- a/tests/check/elements/qtmux.c
+++ b/tests/check/elements/qtmux.c
@@ -28,6 +28,8 @@
 #include <unistd.h>
 #endif
 
+#include <glib/gstdio.h>
+
 #include <gst/check/gstcheck.h>
 #include <gst/pbutils/encoding-profile.h>
 
@@ -830,7 +832,12 @@ test_average_bitrate_custom (const gchar * elementname,
         (guint) gst_util_uint64_scale_round ((guint64) total_bytes,
         (guint64) 8 * GST_SECOND, (guint64) total_duration);
     fail_unless (bitrate == expected);
+    gst_tag_list_free (taglist);
   }
+
+  /* delete file */
+  g_unlink (location);
+  g_free (location);
 }
 
 GST_START_TEST (test_average_bitrate)
diff --git a/tests/check/elements/rgvolume.c b/tests/check/elements/rgvolume.c
index 50dab6a5b..93a336ca2 100644
--- a/tests/check/elements/rgvolume.c
+++ b/tests/check/elements/rgvolume.c
@@ -49,6 +49,8 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
     );
 
+static GstBuffer *test_buffer_new (gfloat value);
+
 /* gstcheck sets up a chain function that appends buffers to a global list.
  * This is our equivalent of that for event handling. */
 static gboolean
@@ -79,6 +81,37 @@ setup_rgvolume (void)
 }
 
 static void
+send_newsegment_and_empty_buffer (void)
+{
+  GstBuffer *buf;
+  GstEvent *ev;
+  GstSegment segment;
+
+  fail_unless (g_list_length (events) == 0);
+
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  ev = gst_event_new_segment (&segment);
+  fail_unless (gst_pad_push_event (mysrcpad, ev),
+      "Pushing newsegment event failed");
+
+  buf = test_buffer_new (0.0);
+  GST_BUFFER_SIZE (buf) = 0;
+  GST_BUFFER_DURATION (buf) = 0;
+  GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET (buf);
+  fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
+
+  fail_unless (g_list_length (events) == 1);
+  fail_unless (events->data == ev);
+  gst_mini_object_unref ((GstMiniObject *) events->data);
+  events = g_list_remove (events, ev);
+
+  fail_unless (g_list_length (buffers) == 1);
+  fail_unless (buffers->data == buf);
+  gst_mini_object_unref ((GstMiniObject *) buffers->data);
+  buffers = g_list_remove (buffers, buf);
+}
+
+static void
 cleanup_rgvolume (GstElement * element)
 {
   GST_DEBUG ("cleanup_rgvolume");
@@ -292,6 +325,8 @@ GST_START_TEST (test_events)
 
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
@@ -340,6 +375,8 @@ GST_START_TEST (test_simple)
       "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
@@ -380,6 +417,8 @@ GST_START_TEST (test_fallback_gain)
       "pre-amp", -6.00, "fallback-gain", -3.00, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, +3.5, GST_TAG_TRACK_PEAK, 1.0,
@@ -423,6 +462,8 @@ GST_START_TEST (test_fallback_track)
       "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, +2.11, GST_TAG_TRACK_PEAK, 1.0, NULL);
@@ -448,6 +489,8 @@ GST_START_TEST (test_fallback_album)
       "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_ALBUM_GAIN, +3.73, GST_TAG_ALBUM_PEAK, 1.0, NULL);
@@ -470,6 +513,8 @@ GST_START_TEST (test_headroom)
       "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, +3.50, GST_TAG_TRACK_PEAK, 1.0, NULL);
@@ -512,6 +557,8 @@ GST_START_TEST (test_reference_level)
       "headroom", +0.00, "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
   set_playing_state (element);
 
+  send_newsegment_and_empty_buffer ();
+
   tag_list = gst_tag_list_new_empty ();
   gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
       GST_TAG_TRACK_GAIN, 0.00, GST_TAG_TRACK_PEAK, 0.2,
diff --git a/tests/check/elements/souphttpsrc.c b/tests/check/elements/souphttpsrc.c
index 6a7e85244..a21ab8ff6 100644
--- a/tests/check/elements/souphttpsrc.c
+++ b/tests/check/elements/souphttpsrc.c
@@ -23,6 +23,8 @@
 # include "config.h"
 #endif
 
+#include <stdlib.h>
+
 #include <glib.h>
 #include <glib/gprintf.h>
 #include <libsoup/soup-address.h>
@@ -444,8 +446,11 @@ souphttpsrc_suite (void)
   TCase *tc_chain, *tc_internet;
 
   g_type_init ();
+
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   if (!g_thread_supported ())
     g_thread_init (NULL);
+#endif
 
   s = suite_create ("souphttpsrc");
   tc_chain = tcase_create ("general");
@@ -453,7 +458,7 @@ souphttpsrc_suite (void)
 
   suite_add_tcase (s, tc_chain);
   run_server (&http_port, &https_port);
-  g_atexit (stop_server);
+  atexit (stop_server);
   tcase_add_test (tc_chain, test_first_buffer_has_offset);
   tcase_add_test (tc_chain, test_redirect_yes);
   tcase_add_test (tc_chain, test_redirect_no);
diff --git a/tests/check/elements/udpsrc.c b/tests/check/elements/udpsrc.c
new file mode 100644
index 000000000..d84334a3e
--- /dev/null
+++ b/tests/check/elements/udpsrc.c
@@ -0,0 +1,120 @@
+/* GStreamer UDP source unit tests
+ * 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.
+ */
+#include <gst/check/gstcheck.h>
+#include <gio/gio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+
+GST_START_TEST (test_udpsrc_empty_packet)
+{
+  GstElement *udpsrc;
+  GSocket *socket;
+  GstPad *sinkpad;
+  int port = 0;
+
+  udpsrc = gst_check_setup_element ("udpsrc");
+  fail_unless (udpsrc != NULL);
+  g_object_set (udpsrc, "port", 0, NULL);
+
+  sinkpad = gst_check_setup_sink_pad_by_name (udpsrc, &sinktemplate, "src");
+  fail_unless (sinkpad != NULL);
+  gst_pad_set_active (sinkpad, TRUE);
+
+  gst_element_set_state (udpsrc, GST_STATE_PLAYING);
+  g_object_get (udpsrc, "port", &port, NULL);
+  GST_INFO ("udpsrc port = %d", port);
+
+  socket = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+      G_SOCKET_PROTOCOL_UDP, NULL);
+
+  if (socket != NULL) {
+    GSocketAddress *sa;
+    GInetAddress *ia;
+    gchar *s;
+
+    ia = g_inet_address_new_loopback (G_SOCKET_FAMILY_IPV4);
+    s = g_inet_address_to_string (ia);
+    GST_LOG ("inet address %s", s);
+    g_free (s);
+    sa = g_inet_socket_address_new (ia, port);
+
+    if (g_socket_send_to (socket, sa, "HeLL0", 0, NULL, NULL) == 0) {
+      GST_INFO ("sent 0 bytes");
+      if (g_socket_send_to (socket, sa, "HeLL0", 6, NULL, NULL) == 6) {
+        GstBuffer *buf;
+        guint len;
+
+        GST_INFO ("sent 6 bytes");
+
+        g_usleep (G_USEC_PER_SEC / 2);
+
+        len = g_list_length (buffers);
+        GST_INFO ("%u buffers", len);
+        fail_unless (len == 1 || len == 2);
+
+        /* last buffer should be our HeLL0 string */
+        buf = GST_BUFFER (g_list_nth_data (buffers, len - 1));
+        fail_unless_equals_int (GST_BUFFER_SIZE (buf), 6);
+        fail_unless_equals_string ((gchar *) GST_BUFFER_DATA (buf), "HeLL0");
+
+        /* if there's another buffer, it should be 0 bytes */
+        if (len == 2) {
+          buf = GST_BUFFER (g_list_nth_data (buffers, 0));
+          fail_unless_equals_int (GST_BUFFER_SIZE (buf), 0);
+        }
+      } else {
+        GST_WARNING ("send_to(6 bytes) failed");
+      }
+    } else {
+      GST_WARNING ("send_to(0 bytes) failed");
+    }
+
+    g_object_unref (sa);
+    g_object_unref (ia);
+  } else {
+    GST_WARNING ("Could not create IPv4 UDP socket for unit test");
+  }
+
+  gst_element_set_state (udpsrc, GST_STATE_NULL);
+
+  gst_check_teardown_pad_by_name (udpsrc, "src");
+  gst_check_teardown_element (udpsrc);
+
+  g_object_unref (socket);
+}
+
+GST_END_TEST;
+
+static Suite *
+udpsrc_suite (void)
+{
+  Suite *s = suite_create ("udpsrc");
+  TCase *tc_chain = tcase_create ("udpsrc");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_udpsrc_empty_packet);
+  return s;
+}
+
+GST_CHECK_MAIN (udpsrc)
diff --git a/tests/files/Makefile.am b/tests/files/Makefile.am
index a0a282937..ae08e1a0f 100644
--- a/tests/files/Makefile.am
+++ b/tests/files/Makefile.am
@@ -7,6 +7,7 @@ EXTRA_DIST = \
 	id3-577468-unsynced-tag.tag \
 	id3-588148-unsynced-v24.tag \
 	pcm16sine.flv \
+	pinknoise-vorbis.mkv \
 	test-cert.pem \
 	test-key.pem
 
diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c
index e8126fad3..fc6d5273e 100644
--- a/tests/icles/equalizer-test.c
+++ b/tests/icles/equalizer-test.c
@@ -179,8 +179,10 @@ main (int argc, char **argv)
   GstPad *eq_sinkpad;
   gchar *uri;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   if (!g_thread_supported ())
     g_thread_init (NULL);
+#endif
 
   /* command line option parsing */
   ctx = g_option_context_new ("FILENAME");
diff --git a/tests/icles/gdkpixbufsink-test.c b/tests/icles/gdkpixbufsink-test.c
index c334326e0..8ada8614b 100644
--- a/tests/icles/gdkpixbufsink-test.c
+++ b/tests/icles/gdkpixbufsink-test.c
@@ -330,8 +330,10 @@ main (int argc, char **argv)
   GOptionContext *ctx;
   GError *opt_err = NULL;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   if (!g_thread_supported ())
     g_thread_init (NULL);
+#endif
 
   gtk_init (&argc, &argv);
 
diff --git a/tests/icles/test-oss4.c b/tests/icles/test-oss4.c
index f15875d7b..233e891fd 100644
--- a/tests/icles/test-oss4.c
+++ b/tests/icles/test-oss4.c
@@ -233,8 +233,10 @@ main (int argc, char **argv)
   GOptionContext *ctx;
   GError *err = NULL;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   if (!g_thread_supported ())
     g_thread_init (NULL);
+#endif
 
   ctx = g_option_context_new ("");
   g_option_context_add_main_entries (ctx, options, NULL);
diff --git a/tests/icles/v4l2src-test.c b/tests/icles/v4l2src-test.c
index 02d005dca..5020c5e7f 100644
--- a/tests/icles/v4l2src-test.c
+++ b/tests/icles/v4l2src-test.c
@@ -491,7 +491,13 @@ main (int argc, char *argv[])
   gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
   loop = g_main_loop_new (NULL, FALSE);
 
-  if (!(input_thread = g_thread_create (read_user, source, TRUE, NULL))) {
+#if !GLIB_CHECK_VERSION (2, 31, 0)
+  input_thread = g_thread_create (read_user, source, TRUE, NULL);
+#else
+  input_thread = g_thread_try_new ("v4l2src-test", read_user, source, NULL);
+#endif
+
+  if (input_thread == NULL) {
     fprintf (stderr, "error: g_thread_create return NULL");
     return -1;
   }
diff --git a/tests/icles/videocrop-test.c b/tests/icles/videocrop-test.c
index 0106e6d41..de8963b3f 100644
--- a/tests/icles/videocrop-test.c
+++ b/tests/icles/videocrop-test.c
@@ -193,8 +193,10 @@ main (int argc, char **argv)
   GstCaps *filter_caps = NULL;
   GList *caps_list, *l;
 
+#if !GLIB_CHECK_VERSION (2, 31, 0)
   if (!g_thread_supported ())
     g_thread_init (NULL);
+#endif
 
   /* command line option parsing */
   ctx = g_option_context_new ("");
author	Sebastian Dröge <sebastian.droege@collabora.co.uk>	2012-01-10 14:32:32 +0100
committer	Sebastian Dröge <sebastian.droege@collabora.co.uk>	2012-01-10 14:32:32 +0100
commit	93e3ed5a869afb79f9658f22ced4691ce475e8a1 (patch)
tree	baaec96daaf22857d61128401c12e2649ef25304
parent	692e00cc00d772fe2b01f518cb9ac88d124807d1 (diff)
parent	2b2c0940f1b7ce8a858a26ad5e246fd645a83830 (diff)