diff --git a/ext/a52dec/Makefile.am b/ext/a52dec/Makefile.am
index d572236cbe..bf56cc9933 100644
--- a/ext/a52dec/Makefile.am
+++ b/ext/a52dec/Makefile.am
@@ -3,12 +3,15 @@ plugin_LTLIBRARIES = libgsta52dec.la
 libgsta52dec_la_SOURCES = gsta52dec.c
 libgsta52dec_la_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
-	$(GST_CFLAGS) \
+        $(GST_BASE_CFLAGS) \
+        $(GST_CFLAGS) \
 	$(ORC_CFLAGS) \
 	$(A52DEC_CFLAGS)
 libgsta52dec_la_LIBADD = \
 	$(GST_PLUGINS_BASE_LIBS) \
-	-lgstaudio-$(GST_MAJORMINOR) \
+        $(GST_BASE_LIBS) \
+        $(GST_LIBS) \
+        -lgstaudio-$(GST_MAJORMINOR) \
 	$(ORC_LIBS) \
 	$(A52DEC_LIBS)
 libgsta52dec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
diff --git a/ext/a52dec/gsta52dec.c b/ext/a52dec/gsta52dec.c
index 558eb9bc10..eef9088eb6 100644
--- a/ext/a52dec/gsta52dec.c
+++ b/ext/a52dec/gsta52dec.c
@@ -21,7 +21,7 @@
  * SECTION:element-a52dec
  *
  * Dolby Digital (AC-3) audio decoder.
- * 
+ *
  * <refsect2>
  * <title>Example launch line</title>
  * |[
@@ -88,17 +88,20 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     );
 
 #define gst_a52dec_parent_class parent_class
-G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_ELEMENT);
+G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_AUDIO_DECODER);
+
+static gboolean gst_a52dec_start (GstAudioDecoder * dec);
+static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
+static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
+static gboolean gst_a52dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+    gint * offset, gint * length);
+static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * buffer);
+static GstFlowReturn gst_a52dec_pre_push (GstAudioDecoder * bdec,
+    GstBuffer ** buffer);
 
 static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstObject * parent,
     GstBuffer * buffer);
-static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstObject * parent,
-    GstBuffer * buf);
-static gboolean gst_a52dec_sink_event (GstPad * pad, GstObject * parent,
-    GstEvent * event);
-static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
-    GstStateChange transition);
-
 static void gst_a52dec_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
 static void gst_a52dec_get_property (GObject * object, guint prop_id,
@@ -132,15 +135,22 @@ gst_a52dec_class_init (GstA52DecClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
+  GstAudioDecoderClass *gstbase_class;
   guint cpuflags;
 
   gobject_class = (GObjectClass *) klass;
   gstelement_class = (GstElementClass *) klass;
+  gstbase_class = (GstAudioDecoderClass *) klass;
 
   gobject_class->set_property = gst_a52dec_set_property;
   gobject_class->get_property = gst_a52dec_get_property;
 
-  gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
+  gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
+  gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
+  gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
+  gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
+  gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
+  gstbase_class->pre_push = GST_DEBUG_FUNCPTR (gst_a52dec_pre_push);
 
   /**
    * GstA52Dec::drc
@@ -159,7 +169,7 @@ gst_a52dec_class_init (GstA52DecClass * klass)
    *
    * Force a particular output channel configuration from the decoder. By default,
    * the channel downmix (if any) is chosen automatically based on the downstream
-   * capabilities of the pipeline. 
+   * capabilities of the pipeline.
    */
   g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
       g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
@@ -213,21 +223,114 @@ gst_a52dec_class_init (GstA52DecClass * klass)
 static void
 gst_a52dec_init (GstA52Dec * a52dec)
 {
-  /* create the sink and src pads */
-  a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
-  gst_pad_set_chain_function (a52dec->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_a52dec_chain));
-  gst_pad_set_event_function (a52dec->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
-  gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
-
-  a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
-  gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
-
   a52dec->request_channels = A52_CHANNEL;
   a52dec->dynamic_range_compression = FALSE;
 
-  gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
+  a52dec->state = NULL;
+  a52dec->samples = NULL;
+
+  /* retrieve and intercept base class chain.
+   * Quite HACKish, but that's dvd specs/caps for you,
+   * since one buffer needs to be split into 2 frames */
+  a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
+  gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
+      GST_DEBUG_FUNCPTR (gst_a52dec_chain));
+}
+
+static gboolean
+gst_a52dec_start (GstAudioDecoder * dec)
+{
+  GstA52Dec *a52dec = GST_A52DEC (dec);
+  GstA52DecClass *klass;
+
+  GST_DEBUG_OBJECT (dec, "start");
+
+  klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
+  a52dec->state = a52_init (klass->a52_cpuflags);
+
+  if (!a52dec->state) {
+    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
+        ("failed to initialize a52 state"));
+    return FALSE;
+  }
+
+  a52dec->samples = a52_samples (a52dec->state);
+  a52dec->bit_rate = -1;
+  a52dec->sample_rate = -1;
+  a52dec->stream_channels = A52_CHANNEL;
+  a52dec->using_channels = A52_CHANNEL;
+  a52dec->level = 1;
+  a52dec->bias = 0;
+  a52dec->flag_update = TRUE;
+
+  /* call upon legacy upstream byte support (e.g. seeking) */
+  gst_audio_decoder_set_byte_time (dec, TRUE);
+
+  return TRUE;
+}
+
+static gboolean
+gst_a52dec_stop (GstAudioDecoder * dec)
+{
+  GstA52Dec *a52dec = GST_A52DEC (dec);
+
+  GST_DEBUG_OBJECT (dec, "stop");
+
+  a52dec->samples = NULL;
+  if (a52dec->state) {
+    a52_free (a52dec->state);
+    a52dec->state = NULL;
+  }
+  if (a52dec->pending_tags) {
+    gst_tag_list_free (a52dec->pending_tags);
+    a52dec->pending_tags = NULL;
+  }
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
+    gint * _offset, gint * len)
+{
+  GstA52Dec *a52dec;
+  const guint8 *data;
+  gint av, size;
+  gint length = 0, flags, sample_rate, bit_rate;
+  GstFlowReturn result = GST_FLOW_EOS;
+
+  a52dec = GST_A52DEC (bdec);
+
+  size = av = gst_adapter_available (adapter);
+  data = (const guint8 *) gst_adapter_map (adapter, av);
+
+  /* find and read header */
+  bit_rate = a52dec->bit_rate;
+  sample_rate = a52dec->sample_rate;
+  flags = 0;
+  while (av >= 7) {
+    length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);
+
+    if (length == 0) {
+      /* shift window to re-find sync */
+      data++;
+      size--;
+    } else if (length <= size) {
+      GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
+      result = GST_FLOW_OK;
+      break;
+    } else {
+      GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
+          length, size);
+      break;
+    }
+  }
+  gst_adapter_unmap (adapter);
+
+  *_offset = av - size;
+  *len = length;
+
+  return result;
 }
 
 static gint
@@ -310,119 +413,15 @@ gst_a52dec_channels (int flags, GstAudioChannelPosition * pos)
   return chans;
 }
 
-static void
-clear_queued (GstA52Dec * dec)
-{
-  g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
-  g_list_free (dec->queued);
-  dec->queued = NULL;
-}
-
-static GstFlowReturn
-flush_queued (GstA52Dec * dec)
-{
-  GstFlowReturn ret = GST_FLOW_OK;
-
-  while (dec->queued) {
-    GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
-
-    GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
-        GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
-        GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
-    /* iterate ouput queue an push downstream */
-    ret = gst_pad_push (dec->srcpad, buf);
-
-    dec->queued = g_list_delete_link (dec->queued, dec->queued);
-  }
-  return ret;
-}
-
-static GstFlowReturn
-gst_a52dec_drain (GstA52Dec * dec)
-{
-  GstFlowReturn ret = GST_FLOW_OK;
-
-  if (dec->segment.rate < 0.0) {
-    /* if we have some queued frames for reverse playback, flush
-     * them now */
-    ret = flush_queued (dec);
-  }
-  return ret;
-}
-
-static GstFlowReturn
-gst_a52dec_push (GstA52Dec * a52dec,
-    GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
-{
-  GstBuffer *buf;
-  int chans, n, c;
-  GstFlowReturn result;
-  sample_t *data;
-
-  flags &= (A52_CHANNEL_MASK | A52_LFE);
-  if (!(chans = gst_a52dec_channels (flags, NULL)))
-    goto no_channels;
-
-  buf = gst_buffer_new_allocate (NULL, 256 * chans * (SAMPLE_WIDTH / 8), 0);
-
-  data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
-  for (n = 0; n < 256; n++) {
-    for (c = 0; c < chans; c++) {
-      data[n * chans + c] = samples[c * 256 + n];
-    }
-  }
-  gst_audio_reorder_channels (data, 256 * chans * (SAMPLE_WIDTH / 8),
-      (SAMPLE_WIDTH == 64) ? GST_AUDIO_FORMAT_F64 : GST_AUDIO_FORMAT_F32, chans,
-      a52dec->from, a52dec->to);
-  gst_buffer_unmap (buf, data, -1);
-
-  GST_BUFFER_TIMESTAMP (buf) = timestamp;
-  GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
-
-  result = GST_FLOW_OK;
-  if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
-              a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
-    /* set discont when needed */
-    if (a52dec->discont) {
-      GST_LOG_OBJECT (a52dec, "marking DISCONT");
-      GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
-      a52dec->discont = FALSE;
-    }
-
-    if (a52dec->segment.rate > 0.0) {
-      GST_DEBUG_OBJECT (a52dec,
-          "Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
-          GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
-          GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
-      result = gst_pad_push (srcpad, buf);
-    } else {
-      /* reverse playback, queue frame till later when we get a discont. */
-      GST_DEBUG_OBJECT (a52dec, "queued frame");
-      a52dec->queued = g_list_prepend (a52dec->queued, buf);
-    }
-  }
-  return result;
-
-  /* ERRORS */
-no_channels:
-  {
-    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
-        ("invalid channel flags: %d", flags));
-    return GST_FLOW_ERROR;
-  }
-}
-
 static gboolean
-gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
+gst_a52dec_reneg (GstA52Dec * a52dec)
 {
   gint channels;
   GstCaps *caps = NULL;
   gboolean result = FALSE;
+  GstAudioChannelPosition from[6], to[6];
 
-  channels = gst_a52dec_channels (a52dec->using_channels, a52dec->from);
+  channels = gst_a52dec_channels (a52dec->using_channels, from);
 
   if (!channels)
     goto done;
@@ -430,8 +429,10 @@ gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
   GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
       channels, a52dec->sample_rate);
 
-  memcpy (a52dec->to, a52dec->from, sizeof (a52dec->from));
-  gst_audio_channel_positions_to_valid_order (a52dec->to, channels);
+  memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
+  gst_audio_channel_positions_to_valid_order (to, channels);
+  gst_audio_get_channel_reorder_map (channels, from, to,
+      a52dec->channel_reorder_map);
 
   caps = gst_caps_new_simple ("audio/x-raw",
       "format", G_TYPE_STRING, SAMPLE_FORMAT,
@@ -441,15 +442,13 @@ gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
 
   if (channels > 1) {
     guint64 channel_mask = 0;
-    gint i;
 
-    for (i = 0; i < channels; i++)
-      channel_mask |= G_GUINT64_CONSTANT (1) << a52dec->to[i];
+    gst_audio_channel_positions_to_mask (to, channels, &channel_mask);
     gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
         NULL);
   }
 
-  if (!gst_pad_set_caps (pad, caps))
+  if (!gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (a52dec), caps))
     goto done;
 
   result = TRUE;
@@ -460,113 +459,69 @@ done:
   return result;
 }
 
-static gboolean
-gst_a52dec_sink_setcaps (GstA52Dec * a52dec, GstCaps * caps)
-{
-  GstStructure *structure;
-
-  structure = gst_caps_get_structure (caps, 0);
-
-  if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
-    a52dec->dvdmode = TRUE;
-  else
-    a52dec->dvdmode = FALSE;
-
-  return TRUE;
-}
-
-static gboolean
-gst_a52dec_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
-{
-  GstA52Dec *a52dec = GST_A52DEC (parent);
-  gboolean ret = FALSE;
-
-  GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_CAPS:
-    {
-      GstCaps *caps;
-
-      gst_event_parse_caps (event, &caps);
-
-      ret = gst_a52dec_sink_setcaps (a52dec, caps);
-      gst_event_unref (event);
-      break;
-    }
-    case GST_EVENT_SEGMENT:
-    {
-      GstSegment seg;
-
-      gst_event_copy_segment (event, &seg);
-
-      /* drain queued buffers before activating the segment so that we can clip
-       * against the old segment first */
-      gst_a52dec_drain (a52dec);
-
-      if (seg.format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (seg.start)) {
-        GST_WARNING ("No time in newsegment event %p (format is %s)",
-            event, gst_format_get_name (seg.format));
-        gst_event_unref (event);
-        a52dec->sent_segment = FALSE;
-        /* set some dummy values, FIXME: do proper conversion */
-        a52dec->time = seg.start = seg.position = 0;
-        seg.format = GST_FORMAT_TIME;
-        seg.stop = -1;
-      } else {
-        a52dec->time = seg.start;
-        a52dec->sent_segment = TRUE;
-        GST_DEBUG_OBJECT (a52dec, "Pushing segment %" GST_SEGMENT_FORMAT, &seg);
-
-        ret = gst_pad_push_event (a52dec->srcpad, event);
-      }
-      a52dec->segment = seg;
-      break;
-    }
-    case GST_EVENT_TAG:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_EOS:
-      gst_a52dec_drain (a52dec);
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_START:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      if (a52dec->cache) {
-        gst_buffer_unref (a52dec->cache);
-        a52dec->cache = NULL;
-      }
-      clear_queued (a52dec);
-      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-    default:
-      ret = gst_pad_push_event (a52dec->srcpad, event);
-      break;
-  }
-
-  return ret;
-}
-
 static void
 gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
 {
   GstTagList *taglist;
 
-  taglist = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
-      GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
+  taglist = gst_tag_list_new_empty ();
+  gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
+      (guint) a52dec->bit_rate, NULL);
 
-  gst_pad_push_event (GST_PAD (a52dec->srcpad), gst_event_new_tag (taglist));
+  if (a52dec->pending_tags) {
+    gst_tag_list_free (a52dec->pending_tags);
+    a52dec->pending_tags = NULL;
+  }
+
+  a52dec->pending_tags = taglist;
 }
 
 static GstFlowReturn
-gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
-    guint length, gint flags, gint sample_rate, gint bit_rate)
+gst_a52dec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer)
 {
+  GstA52Dec *a52dec = GST_A52DEC (bdec);
+
+  if (G_UNLIKELY (a52dec->pending_tags)) {
+    gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (a52dec),
+        gst_event_new_tag (a52dec->pending_tags));
+    a52dec->pending_tags = NULL;
+  }
+
+  return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
+{
+  GstA52Dec *a52dec;
   gint channels, i;
   gboolean need_reneg = FALSE;
+  gint chans;
+  gint length = 0, flags, sample_rate, bit_rate;
+  guint8 *data;
+  gsize size;
+  GstFlowReturn result = GST_FLOW_OK;
+  GstBuffer *outbuf;
+  const gint num_blocks = 6;
+
+  a52dec = GST_A52DEC (bdec);
+
+  /* no fancy draining */
+  if (G_UNLIKELY (!buffer))
+    return GST_FLOW_OK;
+
+  /* parsed stuff already, so this should work out fine */
+  data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+  g_assert (size >= 7);
+
+  /* re-obtain some sync header info,
+   * should be same as during _parse and could also be cached there,
+   * but anyway ... */
+  bit_rate = a52dec->bit_rate;
+  sample_rate = a52dec->sample_rate;
+  flags = 0;
+  length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
+  g_assert (length == size);
 
   /* update stream information, renegotiate or re-streaminfo if needed */
   need_reneg = FALSE;
@@ -585,8 +540,8 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
   }
 
   /* If we haven't had an explicit number of channels chosen through properties
-   * at this point, choose what to downmix to now, based on what the peer will 
-   * accept - this allows a52dec to do downmixing in preference to a 
+   * at this point, choose what to downmix to now, based on what the peer will
+   * accept - this allows a52dec to do downmixing in preference to a
    * downstream element such as audioconvert.
    */
   if (a52dec->request_channels != A52_CHANNEL) {
@@ -596,7 +551,7 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
 
     a52dec->flag_update = FALSE;
 
-    caps = gst_pad_get_allowed_caps (a52dec->srcpad);
+    caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
     if (caps && gst_caps_get_size (caps) > 0) {
       GstCaps *copy = gst_caps_copy_nth (caps, 0);
       GstStructure *structure = gst_caps_get_structure (copy, 0);
@@ -610,13 +565,13 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
         A52_3F2R | A52_LFE,
       };
 
-      /* Prefer the original number of channels, but fixate to something 
+      /* Prefer the original number of channels, but fixate to something
        * preferred (first in the caps) downstream if possible.
        */
       gst_structure_fixate_field_nearest_int (structure, "channels",
           flags ? gst_a52dec_channels (flags, NULL) : 6);
-      gst_structure_get_int (structure, "channels", &channels);
-      if (channels <= 6)
+      if (gst_structure_get_int (structure, "channels", &channels)
+          && channels <= 6)
         flags = a52_channels[channels - 1];
       else
         flags = a52_channels[5];
@@ -632,14 +587,18 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
   } else {
     flags = a52dec->using_channels;
   }
+
   /* process */
   flags |= A52_ADJUST_LEVEL;
   a52dec->level = 1;
   if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
-    GST_WARNING ("a52_frame error");
-    a52dec->discont = TRUE;
-    return GST_FLOW_OK;
+    gst_buffer_unmap (buffer, data, size);
+    GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
+        ("a52_frame error"), result);
+    goto exit;
   }
+  gst_buffer_unmap (buffer, data, size);
+
   channels = flags & (A52_CHANNEL_MASK | A52_LFE);
   if (a52dec->using_channels != channels) {
     need_reneg = TRUE;
@@ -648,57 +607,97 @@ gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
 
   /* negotiate if required */
   if (need_reneg) {
-    GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
+    GST_DEBUG_OBJECT (a52dec,
+        "a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
         a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
-    if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
-      GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
-      return GST_FLOW_ERROR;
-    }
+    if (!gst_a52dec_reneg (a52dec))
+      goto failed_negotiation;
   }
 
   if (a52dec->dynamic_range_compression == FALSE) {
     a52_dynrng (a52dec->state, NULL, NULL);
   }
 
-  /* each frame consists of 6 blocks */
-  for (i = 0; i < 6; i++) {
-    if (a52_block (a52dec->state)) {
-      /* ignore errors but mark a discont */
-      GST_WARNING ("a52_block error %d", i);
-      a52dec->discont = TRUE;
-    } else {
-      GstFlowReturn ret;
+  flags &= (A52_CHANNEL_MASK | A52_LFE);
+  chans = gst_a52dec_channels (flags, NULL);
+  if (!chans)
+    goto invalid_flags;
 
-      /* push on */
-      ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
-          a52dec->samples, a52dec->time);
-      if (ret != GST_FLOW_OK)
-        return ret;
+  /* handle decoded data;
+   * each frame has 6 blocks, one block is 256 samples, ea */
+  outbuf =
+      gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
+
+  data = gst_buffer_map (buffer, &size, NULL, GST_MAP_WRITE);
+  {
+    guint8 *ptr = data;
+    for (i = 0; i < num_blocks; i++) {
+      if (a52_block (a52dec->state)) {
+        /* also marks discont */
+        GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
+            ("error decoding block %d", i), result);
+        if (result != GST_FLOW_OK) {
+          gst_buffer_unmap (outbuf, data, size);
+          goto exit;
+        }
+      } else {
+        gint n, c;
+        gint *reorder_map = a52dec->channel_reorder_map;
+
+        for (n = 0; n < 256; n++) {
+          for (c = 0; c < chans; c++) {
+            ((sample_t *) ptr)[reorder_map[n] * chans + c] =
+                a52dec->samples[c * 256 + n];
+          }
+        }
+      }
+      ptr += 256 * chans * (SAMPLE_WIDTH / 8);
     }
-    a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
   }
+  gst_buffer_unmap (outbuf, data, size);
 
-  return GST_FLOW_OK;
+  result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
+exit:
+  return result;
+
+  /* ERRORS */
+failed_negotiation:
+  {
+    GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
+    return GST_FLOW_ERROR;
+  }
+invalid_flags:
+  {
+    GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
+        ("Invalid channel flags: %d", flags));
+    return GST_FLOW_ERROR;
+  }
+}
+
+static gboolean
+gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+  GstA52Dec *a52dec = GST_A52DEC (bdec);
+  GstStructure *structure;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
+    a52dec->dvdmode = TRUE;
+  else
+    a52dec->dvdmode = FALSE;
+
+  return TRUE;
 }
 
 static GstFlowReturn
 gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
 {
   GstA52Dec *a52dec = GST_A52DEC (parent);
-  GstFlowReturn ret;
+  GstFlowReturn ret = GST_FLOW_OK;
   gint first_access;
 
-  if (GST_BUFFER_IS_DISCONT (buf)) {
-    GST_LOG_OBJECT (a52dec, "received DISCONT");
-    gst_a52dec_drain (a52dec);
-    /* clear cache on discont and mark a discont in the element */
-    if (a52dec->cache) {
-      gst_buffer_unref (a52dec->cache);
-      a52dec->cache = NULL;
-    }
-    a52dec->discont = TRUE;
-  }
-
   if (a52dec->dvdmode) {
     gsize size;
     guint8 data[2];
@@ -724,9 +723,11 @@ gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
 
       subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
       GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
-      ret = gst_a52dec_chain_raw (pad, parent, subbuf);
-      if (ret != GST_FLOW_OK)
+      ret = a52dec->base_chain (pad, parent, subbuf);
+      if (ret != GST_FLOW_OK) {
+        gst_buffer_unref (buf);
         goto done;
+      }
 
       offset += len;
       len = size - offset;
@@ -735,23 +736,23 @@ gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
         subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
         GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
 
-        ret = gst_a52dec_chain_raw (pad, parent, subbuf);
+        ret = a52dec->base_chain (pad, parent, subbuf);
       }
+      gst_buffer_unref (buf);
     } else {
       /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
       subbuf =
           gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
           size - offset);
       GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
-      ret = gst_a52dec_chain_raw (pad, parent, subbuf);
+      ret = a52dec->base_chain (pad, parent, subbuf);
     }
   } else {
     gst_buffer_ref (buf);
-    ret = gst_a52dec_chain_raw (pad, parent, buf);
+    ret = a52dec->base_chain (pad, parent, buf);
   }
 
 done:
-  gst_buffer_unref (buf);
   return ret;
 
 /* ERRORS */
@@ -771,155 +772,6 @@ bad_first_access_parameter:
   }
 }
 
-static GstFlowReturn
-gst_a52dec_chain_raw (GstPad * pad, GstObject * parent, GstBuffer * buf)
-{
-  GstA52Dec *a52dec;
-  guint8 *bdata, *data;
-  gsize bsize, size;
-  gint length = 0, flags, sample_rate, bit_rate;
-  GstFlowReturn result = GST_FLOW_OK;
-
-  a52dec = GST_A52DEC (parent);
-
-  if (!a52dec->sent_segment) {
-    GstSegment segment;
-
-    /* Create a basic segment. Usually, we'll get a new-segment sent by 
-     * another element that will know more information (a demuxer). If we're
-     * just looking at a raw AC3 stream, we won't - so we need to send one
-     * here, but we don't know much info, so just send a minimal TIME 
-     * new-segment event
-     */
-    gst_segment_init (&segment, GST_FORMAT_TIME);
-    gst_pad_push_event (a52dec->srcpad, gst_event_new_segment (&segment));
-    a52dec->sent_segment = TRUE;
-  }
-
-  /* merge with cache, if any. Also make sure timestamps match */
-  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
-    a52dec->time = GST_BUFFER_TIMESTAMP (buf);
-    GST_DEBUG_OBJECT (a52dec,
-        "Received buffer with ts %" GST_TIME_FORMAT " duration %"
-        GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
-        GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-  }
-
-  if (a52dec->cache) {
-    buf = gst_buffer_join (a52dec->cache, buf);
-    a52dec->cache = NULL;
-  }
-  bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
-
-  data = bdata;
-  size = bsize;
-
-  /* find and read header */
-  bit_rate = a52dec->bit_rate;
-  sample_rate = a52dec->sample_rate;
-  flags = 0;
-  while (size >= 7) {
-    length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
-
-    if (length == 0) {
-      /* no sync */
-      data++;
-      size--;
-    } else if (length <= size) {
-      GST_DEBUG ("Sync: %d", length);
-
-      if (flags != a52dec->prev_flags)
-        a52dec->flag_update = TRUE;
-      a52dec->prev_flags = flags;
-
-      result = gst_a52dec_handle_frame (a52dec, data,
-          length, flags, sample_rate, bit_rate);
-      if (result != GST_FLOW_OK) {
-        size = 0;
-        break;
-      }
-      size -= length;
-      data += length;
-    } else {
-      /* not enough data */
-      GST_LOG ("Not enough data available");
-      break;
-    }
-  }
-  gst_buffer_unmap (buf, bdata, bsize);
-
-  /* keep cache */
-  if (length == 0) {
-    GST_LOG ("No sync found");
-  }
-
-  if (size > 0) {
-    a52dec->cache =
-        gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, bsize - size, size);
-  }
-
-  gst_buffer_unref (buf);
-
-  return result;
-}
-
-static GstStateChangeReturn
-gst_a52dec_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
-  GstA52Dec *a52dec = GST_A52DEC (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:{
-      GstA52DecClass *klass;
-
-      klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
-      a52dec->state = a52_init (klass->a52_cpuflags);
-      break;
-    }
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      a52dec->samples = a52_samples (a52dec->state);
-      a52dec->bit_rate = -1;
-      a52dec->sample_rate = -1;
-      a52dec->stream_channels = A52_CHANNEL;
-      a52dec->using_channels = A52_CHANNEL;
-      a52dec->level = 1;
-      a52dec->bias = 0;
-      a52dec->time = 0;
-      a52dec->sent_segment = FALSE;
-      a52dec->flag_update = TRUE;
-      gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      break;
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      a52dec->samples = NULL;
-      if (a52dec->cache) {
-        gst_buffer_unref (a52dec->cache);
-        a52dec->cache = NULL;
-      }
-      clear_queued (a52dec);
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      a52_free (a52dec->state);
-      a52dec->state = NULL;
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
-
 static void
 gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
     GParamSpec * pspec)
diff --git a/ext/a52dec/gsta52dec.h b/ext/a52dec/gsta52dec.h
index 4508a820fc..dd272c14cb 100644
--- a/ext/a52dec/gsta52dec.h
+++ b/ext/a52dec/gsta52dec.h
@@ -23,6 +23,7 @@
 
 #include <gst/gst.h>
 #include <gst/audio/audio.h>
+#include <gst/audio/gstaudiodecoder.h>
 
 G_BEGIN_DECLS
 
@@ -41,27 +42,22 @@ typedef struct _GstA52Dec GstA52Dec;
 typedef struct _GstA52DecClass GstA52DecClass;
 
 struct _GstA52Dec {
-  GstElement     element;
+  GstAudioDecoder element;
 
-  /* pads */
-  GstPad        *sinkpad,
-                *srcpad;
-  GstSegment     segment;
+  GstPadChainFunction base_chain;
 
   gboolean       dvdmode;
-  gboolean       sent_segment;
-  gboolean       discont;
-
   gboolean       flag_update;
   int            prev_flags;
 
+  /* stream properties */
   int            bit_rate;
   int            sample_rate;
   int            stream_channels;
   int            request_channels;
   int            using_channels;
 
-  GstAudioChannelPosition from[6], to[6];
+  gint           channel_reorder_map[6];
 
   sample_t       level;
   sample_t       bias;
@@ -69,15 +65,11 @@ struct _GstA52Dec {
   sample_t      *samples;
   a52_state_t   *state;
 
-  GstBuffer     *cache;
-  GstClockTime   time;
-
-  /* reverse */
-  GList         *queued;
+  GstTagList    *pending_tags;
 };
 
 struct _GstA52DecClass {
-  GstElementClass parent_class;
+  GstAudioDecoderClass parent_class;
 
   guint32 a52_cpuflags;
 };
diff --git a/ext/dvdread/dvdreadsrc.c b/ext/dvdread/dvdreadsrc.c
index 71497a7058..6522854c85 100644
--- a/ext/dvdread/dvdreadsrc.c
+++ b/ext/dvdread/dvdreadsrc.c
@@ -523,15 +523,26 @@ gst_dvd_read_src_goto_title (GstDvdReadSrc * src, gint title, gint angle)
   /* subtitle */
   for (i = 0; i < src->vts_file->vtsi_mat->nr_of_vts_subp_streams; i++) {
     const subp_attr_t *u;
+    const video_attr_t *v;
+    gint sid;
 
     /* subpicture stream present? */
     if (pgc0 != NULL && (pgc0->subp_control[i] & 0x80000000) == 0)
       continue;
 
     u = &src->vts_file->vtsi_mat->vts_subp_attr[i];
+    v = &src->vts_file->vtsi_mat->vts_video_attr;
+
+    sid = i;
+    if (pgc0 != NULL) {
+      if (v->display_aspect_ratio == 0) /* 4:3 */
+        sid = (pgc0->subp_control[i] >> 24) & 0x1f;
+      else if (v->display_aspect_ratio == 3)    /* 16:9 */
+        sid = (pgc0->subp_control[i] >> 8) & 0x1f;
+    }
 
     if (u->type) {
-      t = g_strdup_printf ("subtitle-%d-language", i);
+      t = g_strdup_printf ("subtitle-%d-language", sid);
       lang_code[0] = (u->lang_code >> 8) & 0xff;
       lang_code[1] = u->lang_code & 0xff;
       gst_structure_set (s, t, G_TYPE_STRING, lang_code, NULL);
@@ -541,7 +552,7 @@ gst_dvd_read_src_goto_title (GstDvdReadSrc * src, gint title, gint angle)
     }
 
     GST_INFO_OBJECT (src, "[%02d] Subtitle %02d: lang='%s', type=%d",
-        src->title + 1, i, lang_code, u->type);
+        src->title + 1, sid, lang_code, u->type);
   }
 
   src->title_lang_event_pending =
@@ -1591,7 +1602,8 @@ gst_dvd_read_src_goto_sector (GstDvdReadSrc * src, int angle)
       gint first = src->cur_pgc->cell_playback[cur].first_sector;
       gint last = src->cur_pgc->cell_playback[cur].last_sector;
       GST_DEBUG_OBJECT (src, "Cell %d sector bounds: %d %d", cur, first, last);
-      if (seek_to >= first && seek_to <= last) {
+      /* seeking to 0 should end up at first chapter in any case */
+      if ((seek_to >= first && seek_to <= last) || (seek_to == 0 && i == 0)) {
         GST_DEBUG_OBJECT (src, "Seek target found in chapter %d", i);
         chapter = i;
         goto done;
diff --git a/ext/lame/Makefile.am b/ext/lame/Makefile.am
index 103e68ea8c..03a1dc4f61 100644
--- a/ext/lame/Makefile.am
+++ b/ext/lame/Makefile.am
@@ -2,9 +2,9 @@ plugin_LTLIBRARIES = libgstlame.la
 
 libgstlame_la_SOURCES = gstlamemp3enc.c plugin.c
 libgstlame_la_CFLAGS = -DGST_USE_UNSTABLE_API \
-	$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
-libgstlame_la_LIBADD = $(LAME_LIBS) $(GST_PLUGINS_BASE_LIBS) \
-	-lgstaudio-$(GST_MAJORMINOR) $(GST_LIBS)
+        $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
+libgstlame_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(LAME_LIBS)
 libgstlame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstlame_la_LIBTOOLFLAGS = --tag=disable-static
 
diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
index 099887377c..d0a1813695 100644
--- a/ext/lame/gstlamemp3enc.c
+++ b/ext/lame/gstlamemp3enc.c
@@ -289,6 +289,11 @@ gst_lamemp3enc_start (GstAudioEncoder * enc)
   GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
 
   GST_DEBUG_OBJECT (lame, "start");
+
+  if (!lame->adapter)
+    lame->adapter = gst_adapter_new ();
+  gst_adapter_clear (lame->adapter);
+
   return TRUE;
 }
 
@@ -299,6 +304,11 @@ gst_lamemp3enc_stop (GstAudioEncoder * enc)
 
   GST_DEBUG_OBJECT (lame, "stop");
 
+  if (lame->adapter) {
+    g_object_unref (lame->adapter);
+    lame->adapter = NULL;
+  }
+
   gst_lamemp3enc_release_memory (lame);
   return TRUE;
 }
@@ -334,6 +344,7 @@ gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
         "output samplerate %d is different from incoming samplerate %d",
         out_samplerate, lame->samplerate);
   }
+  lame->out_samplerate = out_samplerate;
 
   version = lame_get_version (lame->lgf);
   if (version == 0)
@@ -362,8 +373,10 @@ gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
       GST_SECOND, lame->samplerate);
   gst_audio_encoder_set_latency (enc, latency, latency);
 
-  if (tags)
+  if (tags) {
     gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
+    gst_tag_list_free (tags);
+  }
 
   return TRUE;
 
@@ -486,6 +499,204 @@ gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
   }
 }
 
+/* **** credits go to mpegaudioparse **** */
+
+static const guint mp3types_bitrates[2][3][16] = {
+  {
+        {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
+        {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
+        {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
+      },
+  {
+        {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
+        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
+        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
+      },
+};
+
+static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
+{22050, 24000, 16000},
+{11025, 12000, 8000}
+};
+
+static inline guint
+mp3_type_frame_length_from_header (GstLameMP3Enc * lame, guint32 header,
+    guint * put_version, guint * put_layer, guint * put_channels,
+    guint * put_bitrate, guint * put_samplerate, guint * put_mode,
+    guint * put_crc)
+{
+  guint length;
+  gulong mode, samplerate, bitrate, layer, channels, padding, crc;
+  gulong version;
+  gint lsf, mpg25;
+
+  if (header & (1 << 20)) {
+    lsf = (header & (1 << 19)) ? 0 : 1;
+    mpg25 = 0;
+  } else {
+    lsf = 1;
+    mpg25 = 1;
+  }
+
+  version = 1 + lsf + mpg25;
+
+  layer = 4 - ((header >> 17) & 0x3);
+
+  crc = (header >> 16) & 0x1;
+
+  bitrate = (header >> 12) & 0xF;
+  bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
+  /* The caller has ensured we have a valid header, so bitrate can't be
+     zero here. */
+  g_assert (bitrate != 0);
+
+  samplerate = (header >> 10) & 0x3;
+  samplerate = mp3types_freqs[lsf + mpg25][samplerate];
+
+  padding = (header >> 9) & 0x1;
+
+  mode = (header >> 6) & 0x3;
+  channels = (mode == 3) ? 1 : 2;
+
+  switch (layer) {
+    case 1:
+      length = 4 * ((bitrate * 12) / samplerate + padding);
+      break;
+    case 2:
+      length = (bitrate * 144) / samplerate + padding;
+      break;
+    default:
+    case 3:
+      length = (bitrate * 144) / (samplerate << lsf) + padding;
+      break;
+  }
+
+  GST_DEBUG_OBJECT (lame, "Calculated mp3 frame length of %u bytes", length);
+  GST_DEBUG_OBJECT (lame, "samplerate = %lu, bitrate = %lu, version = %lu, "
+      "layer = %lu, channels = %lu", samplerate, bitrate, version,
+      layer, channels);
+
+  if (put_version)
+    *put_version = version;
+  if (put_layer)
+    *put_layer = layer;
+  if (put_channels)
+    *put_channels = channels;
+  if (put_bitrate)
+    *put_bitrate = bitrate;
+  if (put_samplerate)
+    *put_samplerate = samplerate;
+  if (put_mode)
+    *put_mode = mode;
+  if (put_crc)
+    *put_crc = crc;
+
+  return length;
+}
+
+static gboolean
+mp3_sync_check (GstLameMP3Enc * lame, unsigned long head)
+{
+  GST_DEBUG_OBJECT (lame, "checking mp3 header 0x%08lx", head);
+  /* if it's not a valid sync */
+  if ((head & 0xffe00000) != 0xffe00000) {
+    GST_WARNING_OBJECT (lame, "invalid sync");
+    return FALSE;
+  }
+  /* if it's an invalid MPEG version */
+  if (((head >> 19) & 3) == 0x1) {
+    GST_WARNING_OBJECT (lame, "invalid MPEG version: 0x%lx", (head >> 19) & 3);
+    return FALSE;
+  }
+  /* if it's an invalid layer */
+  if (!((head >> 17) & 3)) {
+    GST_WARNING_OBJECT (lame, "invalid layer: 0x%lx", (head >> 17) & 3);
+    return FALSE;
+  }
+  /* if it's an invalid bitrate */
+  if (((head >> 12) & 0xf) == 0x0) {
+    GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx."
+        "Free format files are not supported yet", (head >> 12) & 0xf);
+    return FALSE;
+  }
+  if (((head >> 12) & 0xf) == 0xf) {
+    GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
+    return FALSE;
+  }
+  /* if it's an invalid samplerate */
+  if (((head >> 10) & 0x3) == 0x3) {
+    GST_WARNING_OBJECT (lame, "invalid samplerate: 0x%lx", (head >> 10) & 0x3);
+    return FALSE;
+  }
+
+  if ((head & 0x3) == 0x2) {
+    /* Ignore this as there are some files with emphasis 0x2 that can
+     * be played fine. See BGO #537235 */
+    GST_WARNING_OBJECT (lame, "invalid emphasis: 0x%lx", head & 0x3);
+  }
+
+  return TRUE;
+}
+
+/* **** end mpegaudioparse **** */
+
+static GstFlowReturn
+gst_lamemp3enc_finish_frames (GstLameMP3Enc * lame)
+{
+  gint av;
+  guint header;
+  GstFlowReturn result = GST_FLOW_OK;
+
+  /* limited parsing, we don't expect to lose sync here */
+  while ((result == GST_FLOW_OK) &&
+      ((av = gst_adapter_available (lame->adapter)) > 4)) {
+    guint rate, version, layer, size;
+    GstBuffer *mp3_buf;
+    const guint8 *data;
+
+    data = gst_adapter_map (lame->adapter, 4);
+    header = GST_READ_UINT32_BE (data);
+    gst_adapter_unmap (lame->adapter);
+
+    if (!mp3_sync_check (lame, header))
+      goto invalid_header;
+
+    size = mp3_type_frame_length_from_header (lame, header, &version, &layer,
+        NULL, NULL, &rate, NULL, NULL);
+
+    if (G_UNLIKELY (layer != 3 || rate != lame->out_samplerate)) {
+      GST_DEBUG_OBJECT (lame,
+          "unexpected mp3 header with rate %u, version %u, layer %u",
+          rate, version, layer);
+      goto invalid_header;
+    }
+
+    if (size > av) {
+      /* pretty likely to occur when lame is holding back on us */
+      GST_LOG_OBJECT (lame, "frame size %u (> %d)", size, av);
+      break;
+    }
+
+    /* should be ok now */
+    mp3_buf = gst_adapter_take_buffer (lame->adapter, size);
+    /* number of samples for MPEG-1, layer 3 */
+    result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame),
+        mp3_buf, version == 1 ? 1152 : 576);
+  }
+
+exit:
+  return result;
+
+  /* ERRORS */
+invalid_header:
+  {
+    GST_ELEMENT_ERROR (lame, STREAM, ENCODE,
+        ("invalid lame mp3 sync header %08X", header), (NULL));
+    result = GST_FLOW_ERROR;
+    goto exit;
+  }
+}
+
 static GstFlowReturn
 gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
 {
@@ -493,6 +704,7 @@ gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
   gint size;
   guint8 *data;
   GstFlowReturn result = GST_FLOW_OK;
+  gint av;
 
   if (!lame->lgf)
     return GST_FLOW_OK;
@@ -503,17 +715,30 @@ gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
 
   if (size > 0) {
     gst_buffer_unmap (buf, data, size);
-    if (push) {
-      GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
-      result =
-          gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
-    }
+    GST_DEBUG_OBJECT (lame, "collecting final %d bytes", size);
+    gst_adapter_push (lame->adapter, buf);
   } else {
     gst_buffer_unmap (buf, data, 0);
     GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
     gst_buffer_unref (buf);
     result = GST_FLOW_OK;
   }
+
+  if (push) {
+    result = gst_lamemp3enc_finish_frames (lame);
+  } else {
+    /* never mind */
+    gst_adapter_clear (lame->adapter);
+  }
+
+  /* either way, we expect nothing left */
+  if ((av = gst_adapter_available (lame->adapter))) {
+    /* should this be more fatal ?? */
+    GST_WARNING_OBJECT (lame, "unparsed %d bytes left after flushing", av);
+    /* clean up anyway */
+    gst_adapter_clear (lame->adapter);
+  }
+
   return result;
 }
 
@@ -566,8 +791,11 @@ gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
       "to %d bytes of mp3", size, mp3_size);
 
   if (G_LIKELY (mp3_size > 0)) {
+    /* unfortunately lame does not provide frame delineated output,
+     * so collect output and parse into frames ... */
     gst_buffer_unmap (mp3_buf, mp3_data, mp3_size);
-    result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
+    gst_adapter_push (lame->adapter, mp3_buf);
+    result = gst_lamemp3enc_finish_frames (lame);
   } else {
     gst_buffer_unmap (mp3_buf, mp3_data, 0);
     if (mp3_size < 0) {
diff --git a/ext/lame/gstlamemp3enc.h b/ext/lame/gstlamemp3enc.h
index 9bf1225428..d51263e1be 100644
--- a/ext/lame/gstlamemp3enc.h
+++ b/ext/lame/gstlamemp3enc.h
@@ -25,6 +25,7 @@
 
 #include <gst/gst.h>
 #include <gst/audio/gstaudioencoder.h>
+#include <gst/base/gstadapter.h>
 
 G_BEGIN_DECLS
 
@@ -54,6 +55,7 @@ struct _GstLameMP3Enc {
 
   /*< private >*/
   gint samplerate;
+  gint out_samplerate;
   gint num_channels;
 
   /* properties */
@@ -65,6 +67,8 @@ struct _GstLameMP3Enc {
   gboolean mono;
 
   lame_global_flags *lgf;
+
+  GstAdapter *adapter;
 };
 
 struct _GstLameMP3EncClass {
diff --git a/ext/mad/Makefile.am b/ext/mad/Makefile.am
index 85a22ff51d..95117ddd4d 100644
--- a/ext/mad/Makefile.am
+++ b/ext/mad/Makefile.am
@@ -3,11 +3,11 @@ plugin_LTLIBRARIES = libgstmad.la
 libgstmad_la_SOURCES = gstmad.c
 
 libgstmad_la_CFLAGS = \
-	$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
+        $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) \
 	$(MAD_CFLAGS)
 libgstmad_la_LIBADD = \
-	$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
-	-lgstaudio-$(GST_MAJORMINOR) $(MAD_LIBS)
+        $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(MAD_LIBS)
 libgstmad_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstmad_la_LIBTOOLFLAGS = --tag=disable-static
 
diff --git a/ext/mad/gstmad.c b/ext/mad/gstmad.c
index db3cc20b70..a896650807 100644
--- a/ext/mad/gstmad.c
+++ b/ext/mad/gstmad.c
@@ -80,6 +80,7 @@ static gboolean gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
     gint * offset, gint * length);
 static GstFlowReturn gst_mad_handle_frame (GstAudioDecoder * dec,
     GstBuffer * buffer);
+static gboolean gst_mad_event (GstAudioDecoder * dec, GstEvent * event);
 static void gst_mad_flush (GstAudioDecoder * dec, gboolean hard);
 
 static void gst_mad_set_property (GObject * object, guint prop_id,
@@ -102,6 +103,13 @@ gst_mad_class_init (GstMadClass * klass)
   base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
   base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
 
+  base_class->start = GST_DEBUG_FUNCPTR (gst_mad_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_mad_stop);
+  base_class->parse = GST_DEBUG_FUNCPTR (gst_mad_parse);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mad_handle_frame);
+  base_class->flush = GST_DEBUG_FUNCPTR (gst_mad_flush);
+  base_class->event = GST_DEBUG_FUNCPTR (gst_mad_event);
+
   gobject_class->set_property = gst_mad_set_property;
   gobject_class->get_property = gst_mad_get_property;
 
@@ -159,6 +167,7 @@ gst_mad_start (GstAudioDecoder * dec)
   mad_stream_options (&mad->stream, options);
   mad->header.mode = -1;
   mad->header.emphasis = -1;
+  mad->eos = FALSE;
 
   /* call upon legacy upstream byte support (e.g. seeking) */
   gst_audio_decoder_set_byte_time (dec, TRUE);
@@ -269,7 +278,6 @@ gst_mad_check_caps_reset (GstMad * mad)
   }
 }
 
-/* FIXME: does this work properly at all? filesrc ! mad ! pulsesink fails */
 static GstFlowReturn
 gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
     gint * _offset, gint * len)
@@ -281,6 +289,20 @@ gst_mad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
 
   mad = GST_MAD (dec);
 
+  if (mad->eos) {
+    /* This is one steaming hack right there.
+     * mad will not decode the last frame if it is not followed by
+     * a number of 0 bytes, due to some buffer overflow, which can
+     * not be fixed for reasons I did not inquire into, see
+     * http://www.mars.org/mailman/public/mad-dev/2001-May/000262.html
+     */
+    GstBuffer *guard = gst_buffer_new_and_alloc (MAD_BUFFER_GUARD);
+    gst_buffer_memset (guard, 0, 0, MAD_BUFFER_GUARD);
+    GST_DEBUG_OBJECT (mad, "Discreetly stuffing %u zero bytes in the adapter",
+        MAD_BUFFER_GUARD);
+    gst_adapter_push (adapter, guard);
+  }
+
   /* we basically let mad library do parsing,
    * and translate that back to baseclass.
    * if a frame is found (and also decoded), subsequent handle_frame
@@ -472,6 +494,21 @@ gst_mad_flush (GstAudioDecoder * dec, gboolean hard)
   }
 }
 
+static gboolean
+gst_mad_event (GstAudioDecoder * dec, GstEvent * event)
+{
+  GstMad *mad;
+
+  mad = GST_MAD (dec);
+  if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
+    GST_DEBUG_OBJECT (mad, "We got EOS, will pad next time");
+    mad->eos = TRUE;
+  }
+
+  /* Let the base class do its usual thing */
+  return FALSE;
+}
+
 static void
 gst_mad_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
diff --git a/ext/mad/gstmad.h b/ext/mad/gstmad.h
index 1d3aedf261..889ea9ab41 100644
--- a/ext/mad/gstmad.h
+++ b/ext/mad/gstmad.h
@@ -46,7 +46,7 @@ typedef struct _GstMadClass GstMadClass;
 
 struct _GstMad
 {
-  GstAudioDecoder audiodecoder;
+  GstAudioDecoder element;
 
   /* state */
   struct mad_stream stream;
@@ -62,6 +62,8 @@ struct _GstMad
   gint times_pending;
   gboolean caps_set;            /* used to keep track of whether to change/update caps */
 
+  gboolean eos;
+
   /* properties */
   gboolean half;
   gboolean ignore_crc;
@@ -69,7 +71,7 @@ struct _GstMad
 
 struct _GstMadClass
 {
-  GstAudioDecoderClass audiodecoder_class;
+  GstAudioDecoderClass parent_class;
 };
 
 GType                   gst_mad_get_type (void);
diff --git a/gst-libs/gst/Makefile.am b/gst-libs/gst/Makefile.am
index 2df58b3b9b..17887bf366 100644
--- a/gst-libs/gst/Makefile.am
+++ b/gst-libs/gst/Makefile.am
@@ -1 +1 @@
-noinst_HEADERS = gst-i18n-plugin.h gettext.h
+noinst_HEADERS = gst-i18n-plugin.h gettext.h glib-compat-private.h
diff --git a/gst-libs/gst/glib-compat-private.h b/gst-libs/gst/glib-compat-private.h
new file mode 100644
index 0000000000..b9248e6861
--- /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/asfdemux/gstasfdemux.c b/gst/asfdemux/gstasfdemux.c
index ac444e100d..9d7cd4018e 100644
--- a/gst/asfdemux/gstasfdemux.c
+++ b/gst/asfdemux/gstasfdemux.c
@@ -32,6 +32,10 @@
 #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/gstutils.h>
 #include <gst/base/gstbytereader.h>
 #include <gst/riff/riff-media.h>
diff --git a/gst/realmedia/rademux.c b/gst/realmedia/rademux.c
index 82fc3a2758..9551c2e420 100644
--- a/gst/realmedia/rademux.c
+++ b/gst/realmedia/rademux.c
@@ -41,6 +41,10 @@
 #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 "rademux.h"
 #include "rmdemux.h"
 #include "rmutils.h"
diff --git a/gst/realmedia/rdtmanager.c b/gst/realmedia/rdtmanager.c
index 8fb55031eb..6bce4dbaa8 100644
--- a/gst/realmedia/rdtmanager.c
+++ b/gst/realmedia/rdtmanager.c
@@ -56,6 +56,8 @@
 #include "rdtmanager.h"
 #include "rdtjitterbuffer.h"
 
+#include <gst/glib-compat-private.h>
+
 #include <stdio.h>
 
 GST_DEBUG_CATEGORY_STATIC (rdtmanager_debug);
diff --git a/gst/realmedia/rmdemux.c b/gst/realmedia/rmdemux.c
index 2e6abe5219..0fbdee2619 100644
--- a/gst/realmedia/rmdemux.c
+++ b/gst/realmedia/rmdemux.c
@@ -28,6 +28,10 @@
 #  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 "rmdemux.h"
 #include "rmutils.h"
 
diff --git a/po/LINGUAS b/po/LINGUAS
index 41a20ce64d..f176326805 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 ms mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN
+af az bg ca cs da de el en_GB eo es eu fi fr gl hu id it ja lt lv ms mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN
diff --git a/po/eo.po b/po/eo.po
new file mode 100644
index 0000000000..774cfc8d69
--- /dev/null
+++ b/po/eo.po
@@ -0,0 +1,76 @@
+# Esperanto translation for gst-plugins-ugly.
+# Copyright (C) 2011 Free Software Foundation, Inc.
+# This file is distributed under the same license as the gst-plugins-ugly package.
+# Kristjan SCHMIDT <kristjan.schmidt@googlemail.com>, 2011.
+#
+msgid ""
+msgstr ""
+"Project-Id-Version: gst-plugins-ugly 0.10.17.2\n"
+"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
+"POT-Creation-Date: 2011-12-10 15:17+0000\n"
+"PO-Revision-Date: 2011-06-04 22:24+0200\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"
+
+msgid "Could not read from CD."
+msgstr "Ne eblis legi de la KD."
+
+msgid "Could not open CD device for reading."
+msgstr "Ne eblis malfermi la KD-aparaton por legi."
+
+msgid "Disc is not an Audio CD."
+msgstr "La disko ne estas son-KD."
+
+msgid "Could not open DVD"
+msgstr "Ne eblis malfermi la DVD-n"
+
+#, c-format
+msgid "Could not open DVD title %d"
+msgstr "Ne eblis malfermi la titolon \"%d\" de la DVD"
+
+#, c-format
+msgid "Failed to go to chapter %d of DVD title %d"
+msgstr "Fiaksi iri al la ĉapitro %d de la DVD-titolo %d"
+
+#, c-format
+msgid ""
+"Could not open DVD title %d. Interactive titles are not supported by this "
+"element"
+msgstr ""
+"Ne eblis malfermi la DVD-titolon \"%d\". Interagaj titoloj ne estas "
+"suptenata de tiu elemento"
+
+msgid ""
+"Could not read DVD. This may be because the DVD is encrypted and a DVD "
+"decryption library is not installed."
+msgstr ""
+"Ne eblis legi la DVD-n. Eble la DVD estas ĉifrita sed biblioteko por DVD-"
+"malĉifrado ne estas instalite."
+
+msgid "Could not read DVD."
+msgstr "Ne eblis legi la DVD-n."
+
+msgid "Failed to configure LAME encoder. Check your encoding parameters."
+msgstr "Fiaskis agordi la LAME-kodilon. Kontrolu viajn kodad-parametrojn."
+
+#, c-format
+msgid ""
+"The requested bitrate %d kbit/s for property '%s' is not allowed. The "
+"bitrate was changed to %d kbit/s."
+msgstr ""
+"La petita bitrapido je %d kBit/s ne estas permesata por la atributo \"%s\". "
+"La bitrapido estas ŝanĝite al %d kBit/s."
+
+msgid "Failed to configure TwoLAME encoder. Check your encoding parameters."
+msgstr "Fiaskis agordi la TwoLAME-kodilon. Kontrolu viajn kodad-parametrojn."
+
+msgid "This stream contains no data."
+msgstr "Tiu fluo enhavas neniun datumon."
+
+msgid "Internal data stream error."
+msgstr "Interna datum-flu-eraro."