Merge branch 'master' into 0.11

Conflicts:
	configure.ac
	ext/kate/gstkateenc.c
	gst/colorspace/colorspace.c
	gst/mpegvideoparse/mpegvideoparse.c

23 files changed, 730 insertions, 336 deletions

diff --git a/ext/Makefile.am b/ext/Makefile.am
index dc62386ca..a1636f690 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -58,6 +58,12 @@ else
 CELT_DIR=
 endif
 
+if USE_CHROMAPRINT
+CHROMAPRINT_DIR=chromaprint
+else
+CHROMAPRINT_DIR=
+endif
+
 if USE_COG
 COG_DIR=cog
 else
@@ -397,6 +403,7 @@ SUBDIRS=\
 	$(BZ2_DIR) \
 	$(CDAUDIO_DIR) \
 	$(CELT_DIR) \
+	$(CHROMAPRINT_DIR) \
 	$(COG_DIR) \
 	$(CURL_DIR) \
 	$(DC1394_DIR) \
@@ -456,6 +463,7 @@ DIST_SUBDIRS = \
 	bz2 \
 	cdaudio \
 	celt \
+	chromaprint \
 	cog \
 	curl \
 	dc1394 \
diff --git a/ext/chromaprint/Makefile.am b/ext/chromaprint/Makefile.am
new file mode 100644
index 000000000..115d8c298
--- /dev/null
+++ b/ext/chromaprint/Makefile.am
@@ -0,0 +1,14 @@
+plugin_LTLIBRARIES = libgstchromaprint.la
+
+libgstchromaprint_la_SOURCES = gstchromaprint.c gstchromaprint.h
+
+libgstchromaprint_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) \
+	$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
+	$(CHROMAPRINT_CFLAGS)
+libgstchromaprint_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(GST_LIBS) \
+	$(CHROMAPRINT_LIBS)
+libgstchromaprint_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstchromaprint_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = gstchromaprint.h
diff --git a/ext/chromaprint/gstchromaprint.c b/ext/chromaprint/gstchromaprint.c
new file mode 100644
index 000000000..c0a129301
--- /dev/null
+++ b/ext/chromaprint/gstchromaprint.c
@@ -0,0 +1,317 @@
+/* GStreamer chromaprint audio fingerprinting element
+ * Copyright (C) 2006 M. Derezynski
+ * Copyright (C) 2008 Eric Buehl
+ * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 2011 Lukáš Lalinský <lalinsky@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-chromaprint
+ *
+ * The chromaprint element calculates an acoustic fingerprint for an
+ * audio stream which can be used to identify a song and look up
+ * further metadata from the <ulink url="http://acoustid.org/">Acoustid</ulink>
+ * and Musicbrainz databases.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch -m uridecodebin uri=file:///path/to/song.ogg ! audioconvert ! chromaprint ! fakesink
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstchromaprint.h"
+
+#define DEFAULT_MAX_DURATION 120
+
+#define PAD_CAPS \
+	"audio/x-raw-int, " \
+        "rate = (int) [ 1, MAX ], " \
+        "channels = (int) [ 1, 2 ], " \
+        "endianness = (int) { BYTE_ORDER  }, " \
+        "width = (int) { 16 }, " \
+        "depth = (int) { 16 }, " \
+	"signed = (boolean) true"
+
+GST_DEBUG_CATEGORY_STATIC (gst_chromaprint_debug);
+#define GST_CAT_DEFAULT gst_chromaprint_debug
+
+enum
+{
+  PROP_0,
+  PROP_FINGERPRINT,
+  PROP_MAX_DURATION
+};
+
+
+GST_BOILERPLATE (GstChromaprint, gst_chromaprint, GstElement,
+    GST_TYPE_AUDIO_FILTER);
+
+static void gst_chromaprint_finalize (GObject * object);
+static void gst_chromaprint_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_chromaprint_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static GstFlowReturn gst_chromaprint_transform_ip (GstBaseTransform * trans,
+    GstBuffer * buf);
+static gboolean gst_chromaprint_event (GstBaseTransform * trans,
+    GstEvent * event);
+
+static void
+gst_chromaprint_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstAudioFilterClass *audio_filter_class = (GstAudioFilterClass *) g_class;
+  GstCaps *caps;
+
+  gst_element_class_set_details_simple (element_class,
+      "Chromaprint fingerprinting element",
+      "Filter/Analyzer/Audio",
+      "Find an audio fingerprint using the Chromaprint library",
+      "Lukáš Lalinský <lalinsky@gmail.com>");
+
+  caps = gst_caps_from_string (PAD_CAPS);
+  gst_audio_filter_class_add_pad_templates (audio_filter_class, caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_chromaprint_class_init (GstChromaprintClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *gstbasetrans_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  gstbasetrans_class = GST_BASE_TRANSFORM_CLASS (klass);
+
+  gobject_class->set_property = gst_chromaprint_set_property;
+  gobject_class->get_property = gst_chromaprint_get_property;
+
+  /* FIXME: do we need this in addition to the tag message ? */
+  g_object_class_install_property (gobject_class, PROP_FINGERPRINT,
+      g_param_spec_string ("fingerprint", "Resulting fingerprint",
+          "Resulting fingerprint", NULL, G_PARAM_READABLE));
+
+  g_object_class_install_property (gobject_class, PROP_MAX_DURATION,
+      g_param_spec_uint ("duration", "Duration limit",
+          "Number of seconds of audio to use for fingerprinting",
+          0, G_MAXUINT, DEFAULT_MAX_DURATION,
+          G_PARAM_READABLE | G_PARAM_WRITABLE));
+
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_chromaprint_finalize);
+
+  gstbasetrans_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_chromaprint_transform_ip);
+  gstbasetrans_class->event = GST_DEBUG_FUNCPTR (gst_chromaprint_event);
+  gstbasetrans_class->passthrough_on_same_caps = TRUE;
+}
+
+static void
+gst_chromaprint_reset (GstChromaprint * chromaprint)
+{
+  if (chromaprint->fingerprint) {
+    chromaprint_dealloc (chromaprint->fingerprint);
+    chromaprint->fingerprint = NULL;
+  }
+
+  chromaprint->nsamples = 0;
+  chromaprint->duration = 0;
+  chromaprint->record = TRUE;
+}
+
+static void
+gst_chromaprint_create_fingerprint (GstChromaprint * chromaprint)
+{
+  GstTagList *tags;
+
+  if (chromaprint->duration <= 3)
+    return;
+
+  GST_DEBUG_OBJECT (chromaprint,
+      "Generating fingerprint based on %d seconds of audio",
+      chromaprint->duration);
+
+  chromaprint_finish (chromaprint->context);
+  chromaprint_get_fingerprint (chromaprint->context, &chromaprint->fingerprint);
+  chromaprint->record = FALSE;
+
+  tags = gst_tag_list_new_full (GST_TAG_CHROMAPRINT_FINGERPRINT,
+      chromaprint->fingerprint, NULL);
+
+  gst_element_found_tags (GST_ELEMENT (chromaprint), tags);
+}
+
+static void
+gst_chromaprint_init (GstChromaprint * chromaprint,
+    GstChromaprintClass * gclass)
+{
+  gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (chromaprint), TRUE);
+
+  chromaprint->context = chromaprint_new (CHROMAPRINT_ALGORITHM_DEFAULT);
+  chromaprint->fingerprint = NULL;
+  chromaprint->max_duration = DEFAULT_MAX_DURATION;
+  gst_chromaprint_reset (chromaprint);
+}
+
+static void
+gst_chromaprint_finalize (GObject * object)
+{
+  GstChromaprint *chromaprint = GST_CHROMAPRINT (object);
+
+  chromaprint->record = FALSE;
+
+  if (chromaprint->context) {
+    chromaprint_free (chromaprint->context);
+    chromaprint->context = NULL;
+  }
+
+  if (chromaprint->fingerprint) {
+    chromaprint_dealloc (chromaprint->fingerprint);
+    chromaprint->fingerprint = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstFlowReturn
+gst_chromaprint_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
+{
+  GstChromaprint *chromaprint = GST_CHROMAPRINT (trans);
+  gint rate = GST_AUDIO_FILTER (chromaprint)->format.rate;
+  gint channels = GST_AUDIO_FILTER (chromaprint)->format.channels;
+  guint nsamples;
+
+  if (G_UNLIKELY (rate <= 0 || channels <= 0))
+    return GST_FLOW_NOT_NEGOTIATED;
+
+  if (!chromaprint->record)
+    return GST_FLOW_OK;
+
+  nsamples = GST_BUFFER_SIZE (buf) / (channels * 2);
+
+  if (nsamples == 0)
+    return GST_FLOW_OK;
+
+  if (chromaprint->nsamples == 0) {
+    chromaprint_start (chromaprint->context, rate, channels);
+  }
+  chromaprint->nsamples += nsamples;
+  chromaprint->duration = chromaprint->nsamples / rate;
+
+  chromaprint_feed (chromaprint->context, GST_BUFFER_DATA (buf),
+      GST_BUFFER_SIZE (buf) / 2);
+
+  if (chromaprint->duration >= chromaprint->max_duration
+      && !chromaprint->fingerprint) {
+    gst_chromaprint_create_fingerprint (chromaprint);
+  }
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_chromaprint_event (GstBaseTransform * trans, GstEvent * event)
+{
+  GstChromaprint *chromaprint = GST_CHROMAPRINT (trans);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_FLUSH_STOP:
+    case GST_EVENT_NEWSEGMENT:
+      GST_DEBUG_OBJECT (trans, "Got %s event, clearing buffer",
+          GST_EVENT_TYPE_NAME (event));
+      gst_chromaprint_reset (chromaprint);
+      break;
+    case GST_EVENT_EOS:
+      if (!chromaprint->fingerprint) {
+        gst_chromaprint_create_fingerprint (chromaprint);
+      }
+      break;
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+static void
+gst_chromaprint_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstChromaprint *chromaprint = GST_CHROMAPRINT (object);
+
+  switch (prop_id) {
+    case PROP_MAX_DURATION:
+      chromaprint->max_duration = g_value_get_uint (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_chromaprint_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstChromaprint *chromaprint = GST_CHROMAPRINT (object);
+
+  switch (prop_id) {
+    case PROP_FINGERPRINT:
+      g_value_set_string (value, chromaprint->fingerprint);
+      break;
+    case PROP_MAX_DURATION:
+      g_value_set_uint (value, chromaprint->max_duration);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  gboolean ret;
+
+  GST_DEBUG_CATEGORY_INIT (gst_chromaprint_debug, "chromaprint",
+      0, "chromaprint element");
+
+  GST_INFO ("libchromaprint %s", chromaprint_get_version ());
+
+  ret = gst_element_register (plugin, "chromaprint", GST_RANK_NONE,
+      GST_TYPE_CHROMAPRINT);
+
+  if (ret) {
+    gst_tag_register (GST_TAG_CHROMAPRINT_FINGERPRINT, GST_TAG_FLAG_META,
+        G_TYPE_STRING, "chromaprint fingerprint", "Chromaprint fingerprint",
+        NULL);
+  }
+
+  return ret;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "chromaprint",
+    "Calculate Chromaprint fingerprint from audio files",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/ext/chromaprint/gstchromaprint.h b/ext/chromaprint/gstchromaprint.h
new file mode 100644
index 000000000..12bad8a13
--- /dev/null
+++ b/ext/chromaprint/gstchromaprint.h
@@ -0,0 +1,77 @@
+/* GStreamer chromaprint audio fingerprinting element
+ * Copyright (C) 2006 M. Derezynski
+ * Copyright (C) 2008 Eric Buehl
+ * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 2011 Lukáš Lalinský <<user@hostname.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 __GST_CHROMAPRINT_H__
+#define __GST_CHROMAPRINT_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstadapter.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/audio/audio.h>
+#include <chromaprint.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_CHROMAPRINT \
+  (gst_chromaprint_get_type())
+#define GST_CHROMAPRINT(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_CHROMAPRINT,GstChromaprint))
+#define GST_CHROMAPRINT_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_CHROMAPRINT,GstChromaprintClass))
+#define GST_IS_CHROMAPRINT(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_CHROMAPRINT))
+#define GST_IS_CHROMAPRINT_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_CHROMAPRINT))
+
+#define GST_TAG_CHROMAPRINT_FINGERPRINT "chromaprint-fingerprint"
+
+typedef struct _GstChromaprint      GstChromaprint;
+typedef struct _GstChromaprintClass GstChromaprintClass;
+
+/**
+ * GstChromaprint:
+ *
+ * Opaque #GstChromaprint element structure
+ */
+struct _GstChromaprint
+{
+  GstAudioFilter element;
+
+  /*< private >*/
+  ChromaprintContext * context;
+  char               * fingerprint;
+  gboolean             record;
+  guint64              nsamples;
+  guint                duration;
+  guint                max_duration;
+};
+
+struct _GstChromaprintClass
+{
+  GstAudioFilterClass parent_class;
+};
+
+GType gst_chromaprint_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_CHROMAPRINT_H__ */
diff --git a/ext/cog/gstcogmse.c b/ext/cog/gstcogmse.c
index 3ce16a37b..ad38d8dfb 100644
--- a/ext/cog/gstcogmse.c
+++ b/ext/cog/gstcogmse.c
@@ -208,6 +208,9 @@ gst_mse_finalize (GObject * object)
   gst_object_unref (fs->sinkpad_test);
   g_mutex_free (fs->lock);
   g_cond_free (fs->cond);
+  gst_buffer_replace (&fs->buffer_ref, NULL);
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
 }
 
 static GstCaps *
@@ -243,7 +246,7 @@ gst_mse_getcaps (GstPad * pad)
   }
 
   if (pad != fs->sinkpad_test) {
-    peercaps = gst_pad_peer_get_caps (fs->sinkpad_ref);
+    peercaps = gst_pad_peer_get_caps (fs->sinkpad_test);
     if (peercaps) {
       icaps = gst_caps_intersect (caps, peercaps);
       gst_caps_unref (caps);
@@ -310,6 +313,7 @@ gst_mse_reset (GstMSE * fs)
   fs->luma_mse_sum = 0;
   fs->chroma_mse_sum = 0;
   fs->n_frames = 0;
+  fs->cancel = FALSE;
 
   if (fs->buffer_ref) {
     gst_buffer_unref (fs->buffer_ref);
@@ -435,9 +439,11 @@ gst_mse_sink_event (GstPad * pad, GstEvent * event)
       break;
     case GST_EVENT_FLUSH_START:
       GST_DEBUG ("flush start");
+      fs->cancel = TRUE;
       break;
     case GST_EVENT_FLUSH_STOP:
       GST_DEBUG ("flush stop");
+      fs->cancel = FALSE;
       break;
     default:
       break;
diff --git a/ext/gme/gstgme.c b/ext/gme/gstgme.c
index 7a70c7a3c..75d9ff423 100644
--- a/ext/gme/gstgme.c
+++ b/ext/gme/gstgme.c
@@ -161,6 +161,8 @@ gst_gme_dec_dispose (GObject * object)
     g_object_unref (gme->adapter);
     gme->adapter = NULL;
   }
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
 }
 
 static GstFlowReturn
diff --git a/ext/gsm/gstgsmdec.c b/ext/gsm/gstgsmdec.c
index 3318bdc77..2bf475f26 100644
--- a/ext/gsm/gstgsmdec.c
+++ b/ext/gsm/gstgsmdec.c
@@ -43,43 +43,16 @@ enum
   ARG_0
 };
 
-static void gst_gsmdec_base_init (gpointer g_class);
-static void gst_gsmdec_class_init (GstGSMDec * klass);
-static void gst_gsmdec_init (GstGSMDec * gsmdec);
-static void gst_gsmdec_finalize (GObject * object);
-
-static gboolean gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps);
-static gboolean gst_gsmdec_sink_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn gst_gsmdec_chain (GstPad * pad, GstBuffer * buf);
-
-static GstElementClass *parent_class = NULL;
+static gboolean gst_gsmdec_start (GstAudioDecoder * dec);
+static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
+static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
+static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
+    GstAdapter * adapter, gint * offset, gint * length);
+static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * in_buf);
 
 /*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */
 
-GType
-gst_gsmdec_get_type (void)
-{
-  static GType gsmdec_type = 0;
-
-  if (!gsmdec_type) {
-    static const GTypeInfo gsmdec_info = {
-      sizeof (GstGSMDecClass),
-      gst_gsmdec_base_init,
-      NULL,
-      (GClassInitFunc) gst_gsmdec_class_init,
-      NULL,
-      NULL,
-      sizeof (GstGSMDec),
-      0,
-      (GInstanceInitFunc) gst_gsmdec_init,
-    };
-
-    gsmdec_type =
-        g_type_register_static (GST_TYPE_ELEMENT, "GstGSMDec", &gsmdec_info, 0);
-  }
-  return gsmdec_type;
-}
-
 #define ENCODED_SAMPLES	160
 
 static GstStaticPadTemplate gsmdec_sink_template =
@@ -101,6 +74,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
         "depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
     );
 
+GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder,
+    GST_TYPE_AUDIO_DECODER);
+
 static void
 gst_gsmdec_base_init (gpointer g_class)
 {
@@ -116,63 +92,60 @@ gst_gsmdec_base_init (gpointer g_class)
 }
 
 static void
-gst_gsmdec_class_init (GstGSMDec * klass)
+gst_gsmdec_class_init (GstGSMDecClass * klass)
 {
-  GObjectClass *gobject_class;
+  GstAudioDecoderClass *base_class;
 
-  gobject_class = (GObjectClass *) klass;
+  base_class = (GstAudioDecoderClass *) klass;
 
-  parent_class = g_type_class_peek_parent (klass);
-
-  gobject_class->finalize = gst_gsmdec_finalize;
+  base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format);
+  base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame);
 
   GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
 }
 
 static void
-gst_gsmdec_init (GstGSMDec * gsmdec)
+gst_gsmdec_init (GstGSMDec * gsmdec, GstGSMDecClass * klass)
 {
-  /* create the sink and src pads */
-  gsmdec->sinkpad =
-      gst_pad_new_from_static_template (&gsmdec_sink_template, "sink");
-  gst_pad_set_setcaps_function (gsmdec->sinkpad, gst_gsmdec_sink_setcaps);
-  gst_pad_set_event_function (gsmdec->sinkpad, gst_gsmdec_sink_event);
-  gst_pad_set_chain_function (gsmdec->sinkpad, gst_gsmdec_chain);
-  gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->sinkpad);
-
-  gsmdec->srcpad =
-      gst_pad_new_from_static_template (&gsmdec_src_template, "src");
-  gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->srcpad);
+}
+
+static gboolean
+gst_gsmdec_start (GstAudioDecoder * dec)
+{
+  GstGSMDec *gsmdec = GST_GSMDEC (dec);
+
+  GST_DEBUG_OBJECT (dec, "start");
 
   gsmdec->state = gsm_create ();
 
-  gsmdec->adapter = gst_adapter_new ();
-  gsmdec->next_of = 0;
-  gsmdec->next_ts = 0;
+  return TRUE;
 }
 
-static void
-gst_gsmdec_finalize (GObject * object)
+static gboolean
+gst_gsmdec_stop (GstAudioDecoder * dec)
 {
-  GstGSMDec *gsmdec;
+  GstGSMDec *gsmdec = GST_GSMDEC (dec);
 
-  gsmdec = GST_GSMDEC (object);
+  GST_DEBUG_OBJECT (dec, "stop");
 
-  g_object_unref (gsmdec->adapter);
   gsm_destroy (gsmdec->state);
 
-  G_OBJECT_CLASS (parent_class)->finalize (object);
+  return TRUE;
 }
 
 static gboolean
-gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
 {
   GstGSMDec *gsmdec;
   GstCaps *srccaps;
   GstStructure *s;
   gboolean ret = FALSE;
+  gint rate;
 
-  gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
+  gsmdec = GST_GSMDEC (dec);
 
   s = gst_caps_get_structure (caps, 0);
   if (s == NULL)
@@ -186,7 +159,9 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
   else
     goto wrong_caps;
 
-  if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
+  gsmdec->needed = 33;
+
+  if (!gst_structure_get_int (s, "rate", &rate)) {
     GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
     goto beach;
   }
@@ -194,21 +169,16 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
   /* MSGSM needs different framing */
   gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
 
-  gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
-      GST_SECOND, gsmdec->rate);
-
   /* Setting up src caps based on the input sample rate. */
   srccaps = gst_caps_new_simple ("audio/x-raw-int",
       "endianness", G_TYPE_INT, G_BYTE_ORDER,
       "signed", G_TYPE_BOOLEAN, TRUE,
       "width", G_TYPE_INT, 16,
       "depth", G_TYPE_INT, 16,
-      "rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
-
-  ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
+      "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL);
 
+  ret = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), srccaps);
   gst_caps_unref (srccaps);
-  gst_object_unref (gsmdec);
 
   return ret;
 
@@ -218,127 +188,66 @@ wrong_caps:
   GST_ERROR_OBJECT (gsmdec, "invalid caps received");
 
 beach:
-  gst_object_unref (gsmdec);
 
   return ret;
 }
 
-static gboolean
-gst_gsmdec_sink_event (GstPad * pad, GstEvent * event)
+static GstFlowReturn
+gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+    gint * offset, gint * length)
 {
-  gboolean res;
-  GstGSMDec *gsmdec;
+  GstGSMDec *gsmdec = GST_GSMDEC (dec);
+  guint size;
+
+  size = gst_adapter_available (adapter);
+  g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
 
-  gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_START:
-      res = gst_pad_push_event (gsmdec->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&gsmdec->segment, GST_FORMAT_UNDEFINED);
-      res = gst_pad_push_event (gsmdec->srcpad, event);
-      break;
-    case GST_EVENT_NEWSEGMENT:
-    {
-      gboolean update;
-      GstFormat format;
-      gdouble rate, arate;
-      gint64 start, stop, time;
-
-      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
-          &start, &stop, &time);
-
-      /* now configure the values */
-      gst_segment_set_newsegment_full (&gsmdec->segment, update,
-          rate, arate, format, start, stop, time);
-
-      /* and forward */
-      res = gst_pad_push_event (gsmdec->srcpad, event);
-      break;
-    }
-    case GST_EVENT_EOS:
-    default:
-      res = gst_pad_push_event (gsmdec->srcpad, event);
-      break;
+  /* WAV49 requires alternating 33 and 32 bytes of input */
+  if (gsmdec->use_wav49) {
+    gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
   }
 
-  gst_object_unref (gsmdec);
+  if (size < gsmdec->needed)
+    return GST_FLOW_UNEXPECTED;
 
-  return res;
+  *offset = 0;
+  *length = gsmdec->needed;
+
+  return GST_FLOW_OK;
 }
 
 static GstFlowReturn
-gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
+gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
 {
   GstGSMDec *gsmdec;
   gsm_byte *data;
   GstFlowReturn ret = GST_FLOW_OK;
-  GstClockTime timestamp;
-  gint needed;
-
-  gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
-
-  timestamp = GST_BUFFER_TIMESTAMP (buf);
-
-  if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
-    gst_adapter_clear (gsmdec->adapter);
-    gsmdec->next_ts = GST_CLOCK_TIME_NONE;
-    /* FIXME, do some good offset */
-    gsmdec->next_of = 0;
-  }
-  gst_adapter_push (gsmdec->adapter, buf);
-
-  needed = 33;
-  /* do we have enough bytes to read a frame */
-  while (gst_adapter_available (gsmdec->adapter) >= needed) {
-    GstBuffer *outbuf;
-
-    /* always the same amount of output samples */
-    outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
-
-    /* If we are not given any timestamp, interpolate from last seen
-     * timestamp (if any). */
-    if (timestamp == GST_CLOCK_TIME_NONE)
-      timestamp = gsmdec->next_ts;
-
-    GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
-
-    /* interpolate in the next run */
-    if (timestamp != GST_CLOCK_TIME_NONE)
-      gsmdec->next_ts = timestamp + gsmdec->duration;
-    timestamp = GST_CLOCK_TIME_NONE;
-
-    GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
-    GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
-    if (gsmdec->next_of != -1)
-      gsmdec->next_of += ENCODED_SAMPLES;
-    GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
-
-    gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
-
-    /* now encode frame into the output buffer */
-    data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
-    if (gsm_decode (gsmdec->state, data,
-            (gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
-      /* invalid frame */
-      GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
-    }
-    gst_adapter_flush (gsmdec->adapter, needed);
-
-    /* WAV49 requires alternating 33 and 32 bytes of input */
-    if (gsmdec->use_wav49)
-      needed = (needed == 33 ? 32 : 33);
-
-    GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
-        GST_BUFFER_SIZE (outbuf),
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
-
-    /* push */
-    ret = gst_pad_push (gsmdec->srcpad, outbuf);
+  GstBuffer *outbuf;
+
+  /* no fancy draining */
+  if (G_UNLIKELY (!buffer))
+    return GST_FLOW_OK;
+
+  gsmdec = GST_GSMDEC (dec);
+
+  /* always the same amount of output samples */
+  outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
+
+  /* now encode frame into the output buffer */
+  data = (gsm_byte *) GST_BUFFER_DATA (buffer);
+  if (gsm_decode (gsmdec->state, data,
+          (gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
+    /* invalid frame */
+    GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
+        ("tried to decode an invalid frame"), ret);
+    if (ret != GST_FLOW_OK)
+      goto exit;
+    gst_buffer_unref (outbuf);
+    outbuf = NULL;
   }
 
-  gst_object_unref (gsmdec);
+  gst_audio_decoder_finish_frame (dec, outbuf, 1);
 
+exit:
   return ret;
 }
diff --git a/ext/gsm/gstgsmdec.h b/ext/gsm/gstgsmdec.h
index 0013aa47e..d3ddec604 100644
--- a/ext/gsm/gstgsmdec.h
+++ b/ext/gsm/gstgsmdec.h
@@ -21,7 +21,7 @@
 #define __GST_GSMDEC_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstadapter.h>
+#include <gst/audio/gstaudiodecoder.h>
 
 #ifdef GSM_HEADER_IN_SUBDIR
 #include <gsm/gsm.h>
@@ -47,28 +47,16 @@ typedef struct _GstGSMDecClass GstGSMDecClass;
 
 struct _GstGSMDec
 {
-  GstElement element;
-
-  /* pads */
-  GstPad *sinkpad, *srcpad;
+  GstAudioDecoder element;
 
   gsm state;
   gint use_wav49;
-  gint64 next_of;
-  GstClockTime next_ts;
-
-  GstAdapter *adapter;
-
-  GstSegment segment;
-
-  gint rate;
-
-  GstClockTime duration;
+  gint needed;
 };
 
 struct _GstGSMDecClass
 {
-  GstElementClass parent_class;
+  GstAudioDecoderClass parent_class;
 };
 
 GType gst_gsmdec_get_type (void);
diff --git a/ext/gsm/gstgsmenc.c b/ext/gsm/gstgsmenc.c
index 434c4b1fa..e8c97c1f0 100644
--- a/ext/gsm/gstgsmenc.c
+++ b/ext/gsm/gstgsmenc.c
@@ -43,39 +43,12 @@ enum
   ARG_0
 };
 
-static void gst_gsmenc_base_init (gpointer g_class);
-static void gst_gsmenc_class_init (GstGSMEnc * klass);
-static void gst_gsmenc_init (GstGSMEnc * gsmenc);
-static void gst_gsmenc_finalize (GObject * object);
-
-static gboolean gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps);
-static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
-
-static GstElementClass *parent_class = NULL;
-
-GType
-gst_gsmenc_get_type (void)
-{
-  static GType gsmenc_type = 0;
-
-  if (!gsmenc_type) {
-    static const GTypeInfo gsmenc_info = {
-      sizeof (GstGSMEncClass),
-      gst_gsmenc_base_init,
-      NULL,
-      (GClassInitFunc) gst_gsmenc_class_init,
-      NULL,
-      NULL,
-      sizeof (GstGSMEnc),
-      0,
-      (GInstanceInitFunc) gst_gsmenc_init,
-    };
-
-    gsmenc_type =
-        g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
-  }
-  return gsmenc_type;
-}
+static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
+static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
+static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
+    GstAudioInfo * info);
+static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
+    GstBuffer * in_buf);
 
 static GstStaticPadTemplate gsmenc_src_template =
 GST_STATIC_PAD_TEMPLATE ("src",
@@ -95,6 +68,9 @@ GST_STATIC_PAD_TEMPLATE ("sink",
         "depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
     );
 
+GST_BOILERPLATE (GstGSMEnc, gst_gsmenc, GstAudioEncoder,
+    GST_TYPE_AUDIO_ENCODER);
+
 static void
 gst_gsmenc_base_init (gpointer g_class)
 {
@@ -110,34 +86,32 @@ gst_gsmenc_base_init (gpointer g_class)
 }
 
 static void
-gst_gsmenc_class_init (GstGSMEnc * klass)
+gst_gsmenc_class_init (GstGSMEncClass * klass)
 {
-  GObjectClass *gobject_class;
+  GstAudioEncoderClass *base_class;
 
-  gobject_class = (GObjectClass *) klass;
+  base_class = (GstAudioEncoderClass *) klass;
 
-  parent_class = g_type_class_peek_parent (klass);
-
-  gobject_class->finalize = gst_gsmenc_finalize;
+  base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
 
   GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
 }
 
 static void
-gst_gsmenc_init (GstGSMEnc * gsmenc)
+gst_gsmenc_init (GstGSMEnc * gsmenc, GstGSMEncClass * klass)
 {
-  gint use_wav49;
+}
 
-  /* create the sink and src pads */
-  gsmenc->sinkpad =
-      gst_pad_new_from_static_template (&gsmenc_sink_template, "sink");
-  gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
-  gst_pad_set_setcaps_function (gsmenc->sinkpad, gst_gsmenc_setcaps);
-  gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
+static gboolean
+gst_gsmenc_start (GstAudioEncoder * enc)
+{
+  GstGSMEnc *gsmenc = GST_GSMENC (enc);
+  gint use_wav49;
 
-  gsmenc->srcpad =
-      gst_pad_new_from_static_template (&gsmenc_src_template, "src");
-  gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
+  GST_DEBUG_OBJECT (enc, "start");
 
   gsmenc->state = gsm_create ();
 
@@ -145,78 +119,69 @@ gst_gsmenc_init (GstGSMEnc * gsmenc)
   use_wav49 = 0;
   gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
 
-  gsmenc->adapter = gst_adapter_new ();
-  gsmenc->next_ts = 0;
+  return TRUE;
 }
 
-static void
-gst_gsmenc_finalize (GObject * object)
+static gboolean
+gst_gsmenc_stop (GstAudioEncoder * enc)
 {
-  GstGSMEnc *gsmenc;
-
-  gsmenc = GST_GSMENC (object);
+  GstGSMEnc *gsmenc = GST_GSMENC (enc);
 
-  g_object_unref (gsmenc->adapter);
+  GST_DEBUG_OBJECT (enc, "stop");
   gsm_destroy (gsmenc->state);
 
-  G_OBJECT_CLASS (parent_class)->finalize (object);
+  return TRUE;
 }
 
 static gboolean
-gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps)
+gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
 {
-  GstGSMEnc *gsmenc;
   GstCaps *srccaps;
 
-  gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
-
   srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
+  gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), srccaps);
 
-  gst_pad_set_caps (gsmenc->srcpad, srccaps);
-
-  gst_object_unref (gsmenc);
+  /* report needs to base class */
+  gst_audio_encoder_set_frame_samples_min (benc, 160);
+  gst_audio_encoder_set_frame_samples_max (benc, 160);
+  gst_audio_encoder_set_frame_max (benc, 1);
 
   return TRUE;
 }
 
-
 static GstFlowReturn
-gst_gsmenc_chain (GstPad * pad, GstBuffer * buf)
+gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
 {
   GstGSMEnc *gsmenc;
   gsm_signal *data;
   GstFlowReturn ret = GST_FLOW_OK;
+  GstBuffer *outbuf;
 
-  gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
+  gsmenc = GST_GSMENC (benc);
 
-  if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
-    gst_adapter_clear (gsmenc->adapter);
+  /* we don't deal with squeezing remnants, so simply discard those */
+  if (G_UNLIKELY (buffer == NULL)) {
+    GST_DEBUG_OBJECT (gsmenc, "no data");
+    goto done;
   }
-  gst_adapter_push (gsmenc->adapter, buf);
-
-  while (gst_adapter_available (gsmenc->adapter) >= 320) {
-    GstBuffer *outbuf;
-
-    outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
 
-    GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
-    GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
-    gsmenc->next_ts += 20 * GST_MSECOND;
-
-    /* encode 160 16-bit samples into 33 bytes */
-    data = (gsm_signal *) gst_adapter_peek (gsmenc->adapter, 320);
-    gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
-    gst_adapter_flush (gsmenc->adapter, 320);
+  if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
+    GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d",
+        GST_BUFFER_SIZE (buffer));
+    ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
+    goto done;
+  }
 
-    gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
+  outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
 
-    GST_DEBUG_OBJECT (gsmenc, "Pushing buffer of size %d",
-        GST_BUFFER_SIZE (outbuf));
+  /* encode 160 16-bit samples into 33 bytes */
+  data = (gsm_signal *) GST_BUFFER_DATA (buffer);
+  gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
 
-    ret = gst_pad_push (gsmenc->srcpad, outbuf);
-  }
+  GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", GST_BUFFER_SIZE (outbuf));
 
-  gst_object_unref (gsmenc);
+  ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
 
+done:
   return ret;
 }
diff --git a/ext/gsm/gstgsmenc.h b/ext/gsm/gstgsmenc.h
index ba3b089b7..28b8e2e29 100644
--- a/ext/gsm/gstgsmenc.h
+++ b/ext/gsm/gstgsmenc.h
@@ -21,7 +21,7 @@
 #define __GST_GSMENC_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstadapter.h>
+#include <gst/audio/gstaudioencoder.h>
 
 #ifdef GSM_HEADER_IN_SUBDIR
 #include <gsm/gsm.h>
@@ -47,20 +47,14 @@ typedef struct _GstGSMEncClass GstGSMEncClass;
 
 struct _GstGSMEnc
 {
-  GstElement element;
-
-  /* pads */
-  GstPad *sinkpad, *srcpad;
-  GstAdapter *adapter;
+  GstAudioEncoder element;
 
   gsm state;
-  GstClockTime next_ts;
-  gboolean firstBuf;
 };
 
 struct _GstGSMEncClass
 {
-  GstElementClass parent_class;
+  GstAudioEncoderClass parent_class;
 };
 
 GType gst_gsmenc_get_type (void);
diff --git a/ext/kate/gstkateenc.c b/ext/kate/gstkateenc.c
index 8ee8b69df..450e8e61e 100644
--- a/ext/kate/gstkateenc.c
+++ b/ext/kate/gstkateenc.c
@@ -924,33 +924,32 @@ gst_kate_enc_chain_text (GstKateEnc * ke, GstBuffer * buf,
         ("kate_encode_set_markup_type: %d", ret));
     rflow = GST_FLOW_ERROR;
   } else {
-    char *text;
-    gsize text_len;
+    const char *text;
+    size_t text_len;
+    gboolean need_unmap = TRUE;
+    kate_float t0 = start / (double) GST_SECOND;
+    kate_float t1 = stop / (double) GST_SECOND;
 
     text = gst_buffer_map (buf, &text_len, NULL, GST_MAP_READ);
-    if (text) {
-      kate_float t0 = start / (double) GST_SECOND;
-      kate_float t1 = stop / (double) GST_SECOND;
-      GST_LOG_OBJECT (ke, "Encoding text: %*.*s (%u bytes) from %f to %f",
-          (int) text_len, (int) text_len, text, text_len, t0, t1);
-
-      ret = kate_encode_text (&ke->k, t0, t1, text, text_len, &kp);
+    if (text == NULL) {
+      text = "";
+      text_len = 0;
+      need_unmap = FALSE;
+    }
 
-      if (G_UNLIKELY (ret < 0)) {
-        GST_ELEMENT_ERROR (ke, STREAM, ENCODE, (NULL),
-            ("Failed to encode text: %d", ret));
-        rflow = GST_FLOW_ERROR;
-      } else {
-        rflow =
-            gst_kate_enc_chain_push_packet (ke, &kp, start, stop - start + 1);
-      }
-    } else {
-      /* FIXME: this should not be an error, we should ignore it and move on */
+    GST_LOG_OBJECT (ke, "Encoding text: %*.*s (%u bytes) from %f to %f",
+        (int) text_len, (int) text_len, GST_BUFFER_DATA (buf),
+        GST_BUFFER_SIZE (buf), t0, t1);
+    ret = kate_encode_text (&ke->k, t0, t1, text, text_len, &kp);
+    if (G_UNLIKELY (ret < 0)) {
       GST_ELEMENT_ERROR (ke, STREAM, ENCODE, (NULL),
-          ("no text in text packet"));
+          ("Failed to encode text: %d", ret));
       rflow = GST_FLOW_ERROR;
+    } else {
+      rflow = gst_kate_enc_chain_push_packet (ke, &kp, start, stop - start + 1);
     }
-    gst_buffer_unmap (buf, text, text_len);
+    if (need_unmap)
+      gst_buffer_unmap (buf, text, text_len);
   }
 
   return rflow;
diff --git a/ext/opencv/gsttemplatematch.c b/ext/opencv/gsttemplatematch.c
index 4f26121db..84640ff3b 100644
--- a/ext/opencv/gsttemplatematch.c
+++ b/ext/opencv/gsttemplatematch.c
@@ -296,6 +296,8 @@ gst_template_match_finalize (GObject * object)
   if (filter->cvTemplateImage) {
     cvReleaseImage (&filter->cvTemplateImage);
   }
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
 }
 
 /* chain function
diff --git a/ext/resindvd/resindvdbin.c b/ext/resindvd/resindvdbin.c
index 5885b238a..8c2d94fd6 100644
--- a/ext/resindvd/resindvdbin.c
+++ b/ext/resindvd/resindvdbin.c
@@ -470,18 +470,16 @@ create_elements (RsnDvdBin * dvdbin)
           RSN_TYPE_STREAM_SELECTOR, "audioselect", "Audio stream selector"))
     return FALSE;
 
-  if (!try_create_piece (dvdbin, DVD_ELEM_AUDDEC, NULL,
-          RSN_TYPE_AUDIODEC, "auddec", "audio decoder"))
+  if (!try_create_piece (dvdbin, DVD_ELEM_AUD_MUNGE, NULL,
+          RSN_TYPE_AUDIOMUNGE, "audioearlymunge", "Audio output filter"))
     return FALSE;
 
-  /* rsnaudiomunge goes after the audio decoding to regulate the stream */
-  if (!try_create_piece (dvdbin, DVD_ELEM_AUD_MUNGE, NULL,
-          RSN_TYPE_AUDIOMUNGE, "audiomunge", "Audio output filter"))
+  if (!try_create_piece (dvdbin, DVD_ELEM_AUDDEC, NULL,
+          RSN_TYPE_AUDIODEC, "auddec", "audio decoder"))
     return FALSE;
 
-  src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "src");
-  sink =
-      gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "sink");
+  src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "src");
+  sink = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "sink");
   if (src == NULL || sink == NULL)
     goto failed_aud_connect;
   if (GST_PAD_LINK_FAILED (gst_pad_link (src, sink)))
@@ -491,7 +489,8 @@ create_elements (RsnDvdBin * dvdbin)
   src = sink = NULL;
 
   src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_SELECT], "src");
-  sink = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "sink");
+  sink =
+      gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "sink");
   if (src == NULL || sink == NULL)
     goto failed_aud_connect;
   if (GST_PAD_LINK_FAILED (gst_pad_link (src, sink)))
@@ -501,7 +500,7 @@ create_elements (RsnDvdBin * dvdbin)
   src = sink = NULL;
 
   /* ghost audio munge output pad onto bin */
-  src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "src");
+  src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "src");
   if (src == NULL)
     goto failed_aud_ghost;
   src_templ = gst_static_pad_template_get (&audio_src_template);
@@ -701,7 +700,7 @@ demux_pad_added (GstElement * element, GstPad * pad, RsnDvdBin * dvdbin)
         gst_element_get_request_pad (dvdbin->pieces[DVD_ELEM_SPU_SELECT],
         "sink_%u");
     skip_mq = TRUE;
-  } else if (can_sink_caps (dvdbin->pieces[DVD_ELEM_AUDDEC], caps)) {
+  } else if (can_sink_caps (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], caps)) {
     GST_LOG_OBJECT (dvdbin, "Found audio pad w/ caps %" GST_PTR_FORMAT, caps);
     dest_pad =
         gst_element_get_request_pad (dvdbin->pieces[DVD_ELEM_AUD_SELECT],
@@ -720,7 +719,7 @@ demux_pad_added (GstElement * element, GstPad * pad, RsnDvdBin * dvdbin)
           ("No MPEG video decoder found"));
     } else {
       GST_ELEMENT_WARNING (dvdbin, STREAM, CODEC_NOT_FOUND, (NULL),
-          ("No MPEG video decoder found"));
+          ("No MPEG audio decoder found"));
     }
   }
 
diff --git a/ext/resindvd/resindvdsrc.c b/ext/resindvd/resindvdsrc.c
index a0059fdaa..e3eba9f91 100644
--- a/ext/resindvd/resindvdsrc.c
+++ b/ext/resindvd/resindvdsrc.c
@@ -269,6 +269,7 @@ rsn_dvdsrc_finalize (GObject * object)
   g_mutex_free (src->dvd_lock);
   g_mutex_free (src->branch_lock);
   g_cond_free (src->still_cond);
+  g_free (src->device);
 
   gst_buffer_replace (&src->alloc_buf, NULL);
   gst_buffer_replace (&src->next_buf, NULL);
diff --git a/ext/resindvd/rsnaudiomunge.c b/ext/resindvd/rsnaudiomunge.c
index 5e6f9cc6f..2b78dfea9 100644
--- a/ext/resindvd/rsnaudiomunge.c
+++ b/ext/resindvd/rsnaudiomunge.c
@@ -155,9 +155,9 @@ rsn_audiomunge_set_caps (GstPad * pad, GstCaps * caps)
   g_return_val_if_fail (munge != NULL, FALSE);
 
   otherpad = (pad == munge->srcpad) ? munge->sinkpad : munge->srcpad;
-  gst_object_unref (munge);
 
   ret = gst_pad_set_caps (otherpad, caps);
+  gst_object_unref (munge);
   return ret;
 }
 
diff --git a/ext/resindvd/rsndec.c b/ext/resindvd/rsndec.c
index 3abc0065c..662e932d5 100644
--- a/ext/resindvd/rsndec.c
+++ b/ext/resindvd/rsndec.c
@@ -247,18 +247,50 @@ _get_decoder_factories (gpointer arg)
   GstPadTemplate *templ = gst_element_class_get_pad_template (klass,
       "sink");
   RsnDecFactoryFilterCtx ctx = { NULL, };
+  GstCaps *raw;
+  gboolean raw_audio;
 
   ctx.desired_caps = gst_pad_template_get_caps (templ);
+
+  raw = gst_caps_from_string ("audio/x-raw-float");
+  raw_audio = gst_caps_can_intersect (raw, ctx.desired_caps);
+  if (raw_audio) {
+    GstCaps *sub = gst_caps_subtract (ctx.desired_caps, raw);
+    ctx.desired_caps = sub;
+  } else {
+    gst_caps_ref (ctx.desired_caps);
+  }
+  gst_caps_unref (raw);
+
   /* Set decoder caps to empty. Will be filled by the factory_filter */
   ctx.decoder_caps = gst_caps_new_empty ();
+  GST_DEBUG ("Finding factories for caps: %" GST_PTR_FORMAT, ctx.desired_caps);
 
   factories = gst_default_registry_feature_filter (
       (GstPluginFeatureFilter) rsndec_factory_filter, FALSE, &ctx);
 
+  /* If these are audio caps, we add audioconvert, which is not a decoder,
+     but allows raw audio to go through relatively unmolested - this will
+     come handy when we have to send placeholder silence to allow preroll
+     for those DVDs which have titles with no audio track. */
+  if (raw_audio) {
+    GstPluginFeature *feature;
+    GST_DEBUG ("These are audio caps, adding audioconvert");
+    feature =
+        gst_default_registry_find_feature ("audioconvert",
+        GST_TYPE_ELEMENT_FACTORY);
+    if (feature) {
+      factories = g_list_append (factories, feature);
+    } else {
+      GST_WARNING ("Could not find feature audioconvert");
+    }
+  }
+
   factories = g_list_sort (factories, (GCompareFunc) sort_by_ranks);
 
   GST_DEBUG ("Available decoder caps %" GST_PTR_FORMAT, ctx.decoder_caps);
   gst_caps_unref (ctx.decoder_caps);
+  gst_caps_unref (ctx.desired_caps);
 
   return factories;
 }
@@ -343,7 +375,7 @@ static GstStaticPadTemplate audio_sink_template =
     GST_STATIC_CAPS ("audio/mpeg,mpegversion=(int)1;"
         "audio/x-private1-lpcm;"
         "audio/x-private1-ac3;" "audio/ac3;" "audio/x-ac3;"
-        "audio/x-private1-dts;")
+        "audio/x-private1-dts; audio/x-raw-float")
     );
 
 static GstStaticPadTemplate audio_src_template = GST_STATIC_PAD_TEMPLATE ("src",
diff --git a/ext/rsvg/gstrsvgoverlay.c b/ext/rsvg/gstrsvgoverlay.c
index 1cbd0990c..9d4ce6025 100644
--- a/ext/rsvg/gstrsvgoverlay.c
+++ b/ext/rsvg/gstrsvgoverlay.c
@@ -123,6 +123,8 @@ static GstStaticPadTemplate data_sink_template =
 GST_BOILERPLATE (GstRsvgOverlay, gst_rsvg_overlay, GstVideoFilter,
     GST_TYPE_VIDEO_FILTER);
 
+static void gst_rsvg_overlay_finalize (GObject * object);
+
 static void
 gst_rsvg_overlay_set_svg_data (GstRsvgOverlay * overlay, const gchar * data,
     gboolean consider_as_filename)
@@ -467,6 +469,7 @@ gst_rsvg_overlay_class_init (GstRsvgOverlayClass * klass)
 
   gobject_class->set_property = gst_rsvg_overlay_set_property;
   gobject_class->get_property = gst_rsvg_overlay_get_property;
+  gobject_class->finalize = gst_rsvg_overlay_finalize;
 
   g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DATA,
       g_param_spec_string ("data", "data", "SVG data.", "",
@@ -542,3 +545,13 @@ gst_rsvg_overlay_init (GstRsvgOverlay * overlay, GstRsvgOverlayClass * klass)
       GST_DEBUG_FUNCPTR (gst_rsvg_overlay_data_sink_event));
   gst_element_add_pad (GST_ELEMENT (overlay), overlay->data_sinkpad);
 }
+
+static void
+gst_rsvg_overlay_finalize (GObject * object)
+{
+  GstRsvgOverlay *overlay = GST_RSVG_OVERLAY (object);
+
+  g_object_unref (overlay->adapter);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
diff --git a/ext/schroedinger/gstschrodec.c b/ext/schroedinger/gstschrodec.c
index 50bb40480..c8fa8336e 100644
--- a/ext/schroedinger/gstschrodec.c
+++ b/ext/schroedinger/gstschrodec.c
@@ -102,7 +102,7 @@ static GstStaticPadTemplate gst_schro_dec_src_template =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV ("{ I420, YUY2, AYUV }"))
+    GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV (GST_SCHRO_YUV_LIST))
     );
 
 GST_BOILERPLATE (GstSchroDec, gst_schro_dec, GstBaseVideoDecoder,
@@ -313,12 +313,25 @@ parse_sequence_header (GstSchroDec * schro_dec, guint8 * data, int size)
   ret = schro_parse_decode_sequence_header (data + 13, size - 13,
       &video_format);
   if (ret) {
-    if (video_format.chroma_format == SCHRO_CHROMA_444) {
-      state->format = GST_VIDEO_FORMAT_AYUV;
-    } else if (video_format.chroma_format == SCHRO_CHROMA_422) {
-      state->format = GST_VIDEO_FORMAT_YUY2;
-    } else if (video_format.chroma_format == SCHRO_CHROMA_420) {
-      state->format = GST_VIDEO_FORMAT_I420;
+    int bit_depth;
+
+    bit_depth = schro_video_format_get_bit_depth (&video_format);
+
+    if (bit_depth == 8) {
+      if (video_format.chroma_format == SCHRO_CHROMA_444) {
+        state->format = GST_VIDEO_FORMAT_AYUV;
+      } else if (video_format.chroma_format == SCHRO_CHROMA_422) {
+        state->format = GST_VIDEO_FORMAT_UYVY;
+      } else if (video_format.chroma_format == SCHRO_CHROMA_420) {
+        state->format = GST_VIDEO_FORMAT_I420;
+      }
+    } else if (bit_depth <= 10) {
+      state->format = GST_VIDEO_FORMAT_v210;
+    } else if (bit_depth <= 16) {
+      state->format = GST_VIDEO_FORMAT_AYUV64;
+    } else {
+      GST_ERROR ("bit depth too large (%d > 16)", bit_depth);
+      state->format = GST_VIDEO_FORMAT_AYUV64;
     }
     state->fps_n = video_format.frame_rate_numerator;
     state->fps_d = video_format.frame_rate_denominator;
diff --git a/ext/schroedinger/gstschroenc.c b/ext/schroedinger/gstschroenc.c
index c2064d3ca..1fb75f98e 100644
--- a/ext/schroedinger/gstschroenc.c
+++ b/ext/schroedinger/gstschroenc.c
@@ -107,7 +107,7 @@ static GstStaticPadTemplate gst_schro_enc_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV ("{ I420, YV12, YUY2, UYVY, AYUV }"))
+    GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV (GST_SCHRO_YUV_LIST))
     );
 
 static GstStaticPadTemplate gst_schro_enc_src_template =
@@ -271,13 +271,18 @@ gst_schro_enc_set_format (GstBaseVideoEncoder * base_video_encoder,
   switch (state->format) {
     case GST_VIDEO_FORMAT_I420:
     case GST_VIDEO_FORMAT_YV12:
+    case GST_VIDEO_FORMAT_Y42B:
       schro_enc->video_format->chroma_format = SCHRO_CHROMA_420;
       break;
     case GST_VIDEO_FORMAT_YUY2:
     case GST_VIDEO_FORMAT_UYVY:
+    case GST_VIDEO_FORMAT_v216:
+    case GST_VIDEO_FORMAT_v210:
       schro_enc->video_format->chroma_format = SCHRO_CHROMA_422;
       break;
     case GST_VIDEO_FORMAT_AYUV:
+    case GST_VIDEO_FORMAT_Y444:
+    case GST_VIDEO_FORMAT_AYUV64:
       schro_enc->video_format->chroma_format = SCHRO_CHROMA_444;
       break;
     case GST_VIDEO_FORMAT_ARGB:
@@ -300,8 +305,24 @@ gst_schro_enc_set_format (GstBaseVideoEncoder * base_video_encoder,
   schro_enc->video_format->aspect_ratio_numerator = state->par_n;
   schro_enc->video_format->aspect_ratio_denominator = state->par_d;
 
-  schro_video_format_set_std_signal_range (schro_enc->video_format,
-      SCHRO_SIGNAL_RANGE_8BIT_VIDEO);
+  switch (state->format) {
+    default:
+      schro_video_format_set_std_signal_range (schro_enc->video_format,
+          SCHRO_SIGNAL_RANGE_8BIT_VIDEO);
+      break;
+    case GST_VIDEO_FORMAT_v210:
+      schro_video_format_set_std_signal_range (schro_enc->video_format,
+          SCHRO_SIGNAL_RANGE_10BIT_VIDEO);
+      break;
+    case GST_VIDEO_FORMAT_v216:
+    case GST_VIDEO_FORMAT_AYUV64:
+      schro_enc->video_format->luma_offset = 64 << 8;
+      schro_enc->video_format->luma_excursion = 219 << 8;
+      schro_enc->video_format->chroma_offset = 128 << 8;
+      schro_enc->video_format->chroma_excursion = 224 << 8;
+      break;
+  }
+
   schro_video_format_set_std_colour_spec (schro_enc->video_format,
       SCHRO_COLOUR_SPEC_HDTV);
 
diff --git a/ext/schroedinger/gstschroutils.c b/ext/schroedinger/gstschroutils.c
index 66514a3d5..99a22c8a4 100644
--- a/ext/schroedinger/gstschroutils.c
+++ b/ext/schroedinger/gstschroutils.c
@@ -72,6 +72,29 @@ gst_schro_buffer_wrap (GstBuffer * buf, GstVideoFormat format, int width,
       frame =
           schro_frame_new_from_data_AYUV (GST_BUFFER_DATA (buf), width, height);
       break;
+    case GST_VIDEO_FORMAT_Y42B:
+      frame =
+          schro_frame_new_from_data_Y42B (GST_BUFFER_DATA (buf), width, height);
+      break;
+    case GST_VIDEO_FORMAT_Y444:
+      frame =
+          schro_frame_new_from_data_Y444 (GST_BUFFER_DATA (buf), width, height);
+      break;
+    case GST_VIDEO_FORMAT_v210:
+      frame =
+          schro_frame_new_from_data_v210 (GST_BUFFER_DATA (buf), width, height);
+      break;
+    case GST_VIDEO_FORMAT_v216:
+      frame =
+          schro_frame_new_from_data_v216 (GST_BUFFER_DATA (buf), width, height);
+      break;
+#ifdef SCHRO_FRAME_FORMAT_AY64
+      /* Added in 1.0.11 */
+    case GST_VIDEO_FORMAT_AYUV64:
+      frame =
+          schro_frame_new_from_data_AY64 (GST_BUFFER_DATA (buf), width, height);
+      break;
+#endif
 #if 0
     case GST_VIDEO_FORMAT_ARGB:
     {
diff --git a/ext/schroedinger/gstschroutils.h b/ext/schroedinger/gstschroutils.h
index 4e8ca2de3..a9924a633 100644
--- a/ext/schroedinger/gstschroutils.h
+++ b/ext/schroedinger/gstschroutils.h
@@ -24,6 +24,12 @@
 #include <gst/video/video.h>
 #include <schroedinger/schro.h>
 
+#ifdef SCHRO_FRAME_FORMAT_AY64
+#define GST_SCHRO_YUV_LIST "{ I420, YV12, YUY2, UYVY, AYUV, Y42B, Y444, v216, v210, AY64 }"
+#else
+#define GST_SCHRO_YUV_LIST "{ I420, YV12, YUY2, UYVY, AYUV, Y42B, Y444 }"
+#endif
+
 SchroFrame *
 gst_schro_buffer_wrap (GstBuffer *buf, GstVideoFormat format, int width,
     int height);
diff --git a/ext/spc/gstspc.c b/ext/spc/gstspc.c
index 2c74a9c56..916718235 100644
--- a/ext/spc/gstspc.c
+++ b/ext/spc/gstspc.c
@@ -169,6 +169,8 @@ gst_spc_dec_dispose (GObject * object)
   }
 
   spc_tag_free (&spc->tag_info);
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
 }
 
 static GstFlowReturn
diff --git a/ext/vp8/gstvp8enc.c b/ext/vp8/gstvp8enc.c
index 7a1832843..ea92a76ce 100644
--- a/ext/vp8/gstvp8enc.c
+++ b/ext/vp8/gstvp8enc.c
@@ -1174,6 +1174,9 @@ gst_vp8_enc_shape_output (GstBaseVideoEncoder * base_video_encoder,
       gst_util_uint64_scale (frame->presentation_frame_number + 1,
       GST_SECOND * state->fps_d, state->fps_n);
 
+  GST_LOG_OBJECT (base_video_encoder, "src ts: %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+
   ret = gst_pad_push (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_encoder), buf);
   if (ret != GST_FLOW_OK) {
     GST_WARNING_OBJECT (encoder, "flow error %d", ret);