diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt
index fb1a18cf7a..8782899d02 100644
--- a/docs/libs/gst-plugins-base-libs-sections.txt
+++ b/docs/libs/gst-plugins-base-libs-sections.txt
@@ -215,7 +215,8 @@ GST_AUDIO_ENCODER_SRC_PAD
 gst_audio_encoder_finish_frame
 gst_audio_encoder_get_audio_info
 gst_audio_encoder_get_frame_max
-gst_audio_encoder_get_frame_samples
+gst_audio_encoder_get_frame_samples_min
+gst_audio_encoder_get_frame_samples_max
 gst_audio_encoder_get_hard_resync
 gst_audio_encoder_get_latency
 gst_audio_encoder_get_lookahead
@@ -224,7 +225,8 @@ gst_audio_encoder_get_perfect_timestamp
 gst_audio_encoder_get_tolerance
 gst_audio_encoder_proxy_getcaps
 gst_audio_encoder_set_frame_max
-gst_audio_encoder_set_frame_samples
+gst_audio_encoder_set_frame_samples_min
+gst_audio_encoder_set_frame_samples_max
 gst_audio_encoder_set_hard_resync
 gst_audio_encoder_set_latency
 gst_audio_encoder_set_lookahead
diff --git a/gst-libs/gst/audio/gstaudiodecoder.c b/gst-libs/gst/audio/gstaudiodecoder.c
index cf92dd47c5..046419831f 100644
--- a/gst-libs/gst/audio/gstaudiodecoder.c
+++ b/gst-libs/gst/audio/gstaudiodecoder.c
@@ -259,6 +259,8 @@ struct _GstAudioDecoderPrivate
   GstClockTime tolerance;
   gboolean plc;
 
+  /* pending serialized sink events, will be sent from finish_frame() */
+  GList *pending_events;
 };
 
 
@@ -375,6 +377,8 @@ gst_audio_decoder_init (GstAudioDecoder * dec)
   dec->priv->adapter_out = gst_adapter_new ();
   g_queue_init (&dec->priv->frames);
 
+  g_static_rec_mutex_init (&dec->stream_lock);
+
   /* property default */
   dec->priv->latency = DEFAULT_LATENCY;
   dec->priv->tolerance = DEFAULT_TOLERANCE;
@@ -390,7 +394,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
 {
   GST_DEBUG_OBJECT (dec, "gst_audio_decoder_reset");
 
-  GST_OBJECT_LOCK (dec);
+  GST_AUDIO_DECODER_STREAM_LOCK (dec);
 
   if (full) {
     dec->priv->active = FALSE;
@@ -409,6 +413,10 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
     }
 
     gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+
+    g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
+    g_list_free (dec->priv->pending_events);
+    dec->priv->pending_events = NULL;
   }
 
   g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
@@ -424,7 +432,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
   dec->priv->discont = TRUE;
   dec->priv->sync_flush = FALSE;
 
-  GST_OBJECT_UNLOCK (dec);
+  GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
 }
 
 static void
@@ -442,6 +450,8 @@ gst_audio_decoder_finalize (GObject * object)
     g_object_unref (dec->priv->adapter_out);
   }
 
+  g_static_rec_mutex_free (&dec->stream_lock);
+
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
@@ -455,6 +465,8 @@ gst_audio_decoder_src_setcaps (GstAudioDecoder * dec, GstCaps * caps)
 
   GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
 
+  GST_AUDIO_DECODER_STREAM_LOCK (dec);
+
   /* parse caps here to check subclass;
    * also makes us aware of output format */
   if (!gst_caps_is_fixed (caps))
@@ -471,6 +483,9 @@ gst_audio_decoder_src_setcaps (GstAudioDecoder * dec, GstCaps * caps)
   if (!gst_audio_info_from_caps (&dec->priv->ctx.info, caps))
     goto refuse_caps;
 
+done:
+  GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
   gst_object_unref (dec);
   return res;
 
@@ -478,8 +493,8 @@ gst_audio_decoder_src_setcaps (GstAudioDecoder * dec, GstCaps * caps)
 refuse_caps:
   {
     GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
-    gst_object_unref (dec);
-    return res;
+    res = FALSE;
+    goto done;
   }
 }
 
@@ -493,6 +508,7 @@ gst_audio_decoder_sink_setcaps (GstAudioDecoder * dec, GstCaps * caps)
 
   GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
 
+  GST_AUDIO_DECODER_STREAM_LOCK (dec);
   /* NOTE pbutils only needed here */
   /* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
 #if 0
@@ -506,6 +522,8 @@ gst_audio_decoder_sink_setcaps (GstAudioDecoder * dec, GstCaps * caps)
   if (klass->set_format)
     res = klass->set_format (dec, caps);
 
+  GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
   return res;
 }
 
@@ -525,7 +543,7 @@ gst_audio_decoder_setup (GstAudioDecoder * dec)
   gst_query_unref (query);
 
   /* normalize to bool */
-  dec->priv->agg = !!res;
+  dec->priv->agg = ! !res;
 }
 
 /* mini aggregator combining output buffers into fewer larger ones,
@@ -677,6 +695,7 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
   gint samples = 0;
   GstClockTime ts, next_ts;
   gsize size;
+  GstFlowReturn ret = GST_FLOW_OK;
 
   /* subclass should know what it is producing by now */
   g_return_val_if_fail (buf == NULL || gst_pad_has_current_caps (dec->srcpad),
@@ -694,6 +713,20 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
   GST_LOG_OBJECT (dec, "accepting %d bytes == %d samples for %d frames",
       buf ? size : -1, buf ? size / ctx->info.bpf : -1, frames);
 
+  GST_AUDIO_DECODER_STREAM_LOCK (dec);
+
+  if (priv->pending_events) {
+    GList *pending_events, *l;
+
+    pending_events = priv->pending_events;
+    priv->pending_events = NULL;
+
+    GST_DEBUG_OBJECT (dec, "Pushing pending events");
+    for (l = priv->pending_events; l; l = l->next)
+      gst_pad_push_event (dec->srcpad, l->data);
+    g_list_free (pending_events);
+  }
+
   /* output shoud be whole number of sample frames */
   if (G_LIKELY (buf && ctx->info.bpf)) {
     if (size % ctx->info.bpf)
@@ -800,7 +833,11 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
     dec->priv->error_count--;
 
 exit:
-  return gst_audio_decoder_output (dec, buf);
+  ret = gst_audio_decoder_output (dec, buf);
+
+  GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
+  return ret;
 
   /* ERRORS */
 wrong_buffer:
@@ -808,7 +845,8 @@ wrong_buffer:
     GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
         ("buffer size %d not a multiple of %d", size, ctx->info.bpf));
     gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
+    ret = GST_FLOW_ERROR;
+    goto exit;
   }
 overflow:
   {
@@ -817,7 +855,8 @@ overflow:
             priv->frames.length), (NULL));
     if (buf)
       gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
+    ret = GST_FLOW_ERROR;
+    goto exit;
   }
 }
 
@@ -1221,6 +1260,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
 
+  GST_AUDIO_DECODER_STREAM_LOCK (dec);
+
   if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
     gint64 samples, ts;
 
@@ -1247,6 +1288,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
   else
     ret = gst_audio_decoder_chain_reverse (dec, buffer);
 
+  GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
   return ret;
 }
 
@@ -1269,6 +1312,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
     {
       GstSegment seg;
 
+      GST_AUDIO_DECODER_STREAM_LOCK (dec);
       gst_event_copy_segment (event, &seg);
 
       if (seg.format == GST_FORMAT_TIME) {
@@ -1296,6 +1340,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
           event = gst_event_new_segment (&seg);
         } else {
           GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
+          GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
           break;
         }
       }
@@ -1339,8 +1384,10 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
 
       /* and follow along with segment */
       dec->segment = seg;
-      gst_pad_push_event (dec->srcpad, event);
+      dec->priv->pending_events =
+          g_list_append (dec->priv->pending_events, event);
       handled = TRUE;
+      GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
       break;
     }
 
@@ -1348,12 +1395,20 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
       break;
 
     case GST_EVENT_FLUSH_STOP:
+      GST_AUDIO_DECODER_STREAM_LOCK (dec);
       /* prepare for fresh start */
       gst_audio_decoder_flush (dec, TRUE);
+
+      g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
+      g_list_free (dec->priv->pending_events);
+      dec->priv->pending_events = NULL;
+      GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
       break;
 
     case GST_EVENT_EOS:
+      GST_AUDIO_DECODER_STREAM_LOCK (dec);
       gst_audio_decoder_drain (dec);
+      GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
       break;
 
     case GST_EVENT_CAPS:
@@ -1393,8 +1448,27 @@ gst_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
   if (!handled)
     handled = gst_audio_decoder_sink_eventfunc (dec, event);
 
-  if (!handled)
-    ret = gst_pad_event_default (pad, event);
+  if (!handled) {
+    /* Forward non-serialized events and EOS/FLUSH_STOP immediately.
+     * For EOS this is required because no buffer or serialized event
+     * will come after EOS and nothing could trigger another
+     * _finish_frame() call.
+     *
+     * For FLUSH_STOP this is required because it is expected
+     * to be forwarded immediately and no buffers are queued anyway.
+     */
+    if (!GST_EVENT_IS_SERIALIZED (event)
+        || GST_EVENT_TYPE (event) == GST_EVENT_EOS
+        || GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
+      ret = gst_pad_event_default (pad, event);
+    } else {
+      GST_AUDIO_DECODER_STREAM_LOCK (dec);
+      dec->priv->pending_events =
+          g_list_append (dec->priv->pending_events, event);
+      GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+      ret = TRUE;
+    }
+  }
 
   GST_DEBUG_OBJECT (dec, "event handled");
 
diff --git a/gst-libs/gst/audio/gstaudiodecoder.h b/gst-libs/gst/audio/gstaudiodecoder.h
index 50fa3adecc..60c6255b37 100644
--- a/gst-libs/gst/audio/gstaudiodecoder.h
+++ b/gst-libs/gst/audio/gstaudiodecoder.h
@@ -20,7 +20,6 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
-
 #ifndef _GST_AUDIO_DECODER_H_
 #define _GST_AUDIO_DECODER_H_
 
@@ -85,6 +84,9 @@ G_BEGIN_DECLS
  */
 #define GST_AUDIO_DECODER_SINK_PAD(obj)        (((GstAudioDecoder *) (obj))->sinkpad)
 
+#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
+#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
+
 typedef struct _GstAudioDecoder GstAudioDecoder;
 typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
 
@@ -146,6 +148,11 @@ struct _GstAudioDecoder
   GstPad         *sinkpad;
   GstPad         *srcpad;
 
+  /* protects all data processing, i.e. is locked
+   * in the chain function, finish_frame and when
+   * processing serialized events */
+  GStaticRecMutex stream_lock;
+
   /* MT-protected (with STREAM_LOCK) */
   GstSegment      segment;
 
diff --git a/gst-libs/gst/audio/gstaudioencoder.c b/gst-libs/gst/audio/gstaudioencoder.c
index 5048905d73..997b0692e8 100644
--- a/gst-libs/gst/audio/gstaudioencoder.c
+++ b/gst-libs/gst/audio/gstaudioencoder.c
@@ -154,6 +154,7 @@
 #include "gstaudioencoder.h"
 #include <gst/base/gstadapter.h>
 #include <gst/audio/audio.h>
+#include <gst/pbutils/descriptions.h>
 
 #include <stdlib.h>
 #include <string.h>
@@ -186,7 +187,7 @@ typedef struct _GstAudioEncoderContext
   GstAudioInfo info;
 
   /* output */
-  gint frame_samples;
+  gint frame_samples_min, frame_samples_max;
   gint frame_max;
   gint lookahead;
   /* MT-protected (with LOCK) */
@@ -238,6 +239,11 @@ struct _GstAudioEncoderPrivate
   gboolean perfect_ts;
   gboolean hard_resync;
   gboolean granule;
+
+  /* pending tags */
+  GstTagList *tags;
+  /* pending serialized sink events, will be sent from finish_frame() */
+  GList *pending_events;
 };
 
 
@@ -380,6 +386,8 @@ gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
 
   enc->priv->adapter = gst_adapter_new ();
 
+  g_static_rec_mutex_init (&enc->stream_lock);
+
   /* property default */
   enc->priv->granule = DEFAULT_GRANULE;
   enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
@@ -394,7 +402,9 @@ gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
 static void
 gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
 {
-  GST_OBJECT_LOCK (enc);
+  GST_AUDIO_ENCODER_STREAM_LOCK (enc);
+
+  GST_LOG_OBJECT (enc, "reset full %d", full);
 
   if (full) {
     enc->priv->active = FALSE;
@@ -402,6 +412,14 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
     enc->priv->bytes_out = 0;
     gst_audio_info_init (&enc->priv->ctx.info);
     memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
+
+    if (enc->priv->tags)
+      gst_tag_list_free (enc->priv->tags);
+    enc->priv->tags = NULL;
+
+    g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
+    g_list_free (enc->priv->pending_events);
+    enc->priv->pending_events = NULL;
   }
 
   gst_segment_init (&enc->segment, GST_FORMAT_TIME);
@@ -415,7 +433,7 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
   enc->priv->samples = 0;
   enc->priv->discont = FALSE;
 
-  GST_OBJECT_UNLOCK (enc);
+  GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
 }
 
 static void
@@ -425,6 +443,8 @@ gst_audio_encoder_finalize (GObject * object)
 
   g_object_unref (enc->priv->adapter);
 
+  g_static_rec_mutex_free (&enc->stream_lock);
+
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
@@ -472,12 +492,41 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
   g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
       GST_FLOW_ERROR);
 
+  GST_AUDIO_ENCODER_STREAM_LOCK (enc);
+
+  if (G_UNLIKELY (enc->priv->tags)) {
+    GstTagList *tags;
+
+    /* add codec info to pending tags */
+    tags = enc->priv->tags;
+    /* no more pending */
+    enc->priv->tags = NULL;
+    gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_CODEC,
+        GST_PAD_CAPS (enc->srcpad));
+    gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
+        GST_PAD_CAPS (enc->srcpad));
+    GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, tags);
+    gst_element_found_tags_for_pad (GST_ELEMENT (enc), enc->srcpad, tags);
+  }
+
   GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
       buf ? gst_buffer_get_size (buf) : -1, samples);
 
   /* mark subclass still alive and providing */
   priv->got_data = TRUE;
 
+  if (priv->pending_events) {
+    GList *pending_events, *l;
+
+    pending_events = priv->pending_events;
+    priv->pending_events = NULL;
+
+    GST_DEBUG_OBJECT (enc, "Pushing pending events");
+    for (l = priv->pending_events; l; l = l->next)
+      gst_pad_push_event (enc->srcpad, l->data);
+    g_list_free (pending_events);
+  }
+
   /* remove corresponding samples from input */
   if (samples < 0)
     samples = (enc->priv->offset / ctx->info.bpf);
@@ -627,6 +676,8 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
   }
 
 exit:
+  GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
+
   return ret;
 
   /* ERRORS */
@@ -637,7 +688,8 @@ overflow:
             samples, priv->offset / ctx->info.bpf), (NULL));
     if (buf)
       gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
+    ret = GST_FLOW_ERROR;
+    goto exit;
   }
 }
 
@@ -670,9 +722,11 @@ gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
     g_assert (priv->offset <= av);
     av -= priv->offset;
 
-    need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->info.bpf : av;
-    GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d",
-        av, need, force);
+    need =
+        ctx->frame_samples_min >
+        0 ? ctx->frame_samples_min * ctx->info.bpf : av;
+    GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d", av, need,
+        force);
 
     if ((need > av) || !av) {
       if (G_UNLIKELY (force)) {
@@ -685,14 +739,19 @@ gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
       priv->force = FALSE;
     }
 
-    /* if we have some extra metadata,
-     * provide for integer multiple of frames to allow for better granularity
-     * of processing */
-    if (ctx->frame_samples > 0 && need) {
-      if (ctx->frame_max > 1)
-        need = need * MIN ((av / need), ctx->frame_max);
-      else if (ctx->frame_max == 0)
-        need = need * (av / need);
+    if (ctx->frame_samples_max > 0)
+      need = MIN (av, ctx->frame_samples_max * ctx->info.bpf);
+
+    if (ctx->frame_samples_min == ctx->frame_samples_max) {
+      /* if we have some extra metadata,
+       * provide for integer multiple of frames to allow for better granularity
+       * of processing */
+      if (ctx->frame_samples_min > 0 && need) {
+        if (ctx->frame_max > 1)
+          need = need * MIN ((av / need), ctx->frame_max);
+        else if (ctx->frame_max == 0)
+          need = need * (av / need);
+      }
     }
 
     if (need) {
@@ -782,6 +841,8 @@ gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
   priv = enc->priv;
   ctx = &enc->priv->ctx;
 
+  GST_AUDIO_ENCODER_STREAM_LOCK (enc);
+
   /* should know what is coming by now */
   if (!ctx->info.bpf)
     goto not_negotiated;
@@ -916,6 +977,9 @@ gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
 
 done:
   GST_LOG_OBJECT (enc, "chain leaving");
+
+  GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
+
   return ret;
 
   /* ERRORS */
@@ -924,7 +988,8 @@ not_negotiated:
     GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
         ("encoder not initialized"));
     gst_buffer_unref (buffer);
-    return GST_FLOW_NOT_NEGOTIATED;
+    ret = GST_FLOW_NOT_NEGOTIATED;
+    goto done;
   }
 wrong_buffer:
   {
@@ -932,7 +997,8 @@ wrong_buffer:
         ("buffer size %d not a multiple of %d", gst_buffer_get_size (buffer),
             ctx->info.bpf));
     gst_buffer_unref (buffer);
-    return GST_FLOW_ERROR;
+    ret = GST_FLOW_ERROR;
+    goto done;
   }
 }
 
@@ -971,6 +1037,8 @@ gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
 
   ctx = &enc->priv->ctx;
 
+  GST_AUDIO_ENCODER_STREAM_LOCK (enc);
+
   GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
 
   if (!gst_caps_is_fixed (caps))
@@ -997,7 +1065,8 @@ gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
     gst_audio_encoder_drain (enc);
 
     /* context defaults */
-    enc->priv->ctx.frame_samples = 0;
+    enc->priv->ctx.frame_samples_min = 0;
+    enc->priv->ctx.frame_samples_max = 0;
     enc->priv->ctx.frame_max = 0;
     enc->priv->ctx.lookahead = 0;
 
@@ -1025,13 +1094,17 @@ gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
     GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
   }
 
+exit:
+
+  GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
+
   return res;
 
   /* ERRORS */
 refuse_caps:
   {
     GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
-    return res;
+    goto exit;
   }
 }
 
@@ -1161,12 +1234,14 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
         break;
       }
 
+      GST_AUDIO_ENCODER_STREAM_LOCK (enc);
       /* finish current segment */
       gst_audio_encoder_drain (enc);
       /* reset partially for new segment */
       gst_audio_encoder_reset (enc, FALSE);
       /* and follow along with segment */
       enc->segment = seg;
+      GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
       break;
     }
 
@@ -1174,18 +1249,46 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
       break;
 
     case GST_EVENT_FLUSH_STOP:
+      GST_AUDIO_ENCODER_STREAM_LOCK (enc);
       /* discard any pending stuff */
       /* TODO route through drain ?? */
       if (!enc->priv->drained && klass->flush)
         klass->flush (enc);
       /* and get (re)set for the sequel */
       gst_audio_encoder_reset (enc, FALSE);
+
+      g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
+      g_list_free (enc->priv->pending_events);
+      enc->priv->pending_events = NULL;
+      GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
+
       break;
 
     case GST_EVENT_EOS:
+      GST_AUDIO_ENCODER_STREAM_LOCK (enc);
       gst_audio_encoder_drain (enc);
+      GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
       break;
 
+    case GST_EVENT_TAG:
+    {
+      GstTagList *tags;
+
+      gst_event_parse_tag (event, &tags);
+      tags = gst_tag_list_copy (tags);
+      gst_event_unref (event);
+      gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
+      gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
+      event = gst_event_new_tag (tags);
+
+      GST_OBJECT_LOCK (enc);
+      enc->priv->pending_events =
+          g_list_append (enc->priv->pending_events, event);
+      GST_OBJECT_UNLOCK (enc);
+      handled = TRUE;
+      break;
+    }
+
     case GST_EVENT_CAPS:
     {
       GstCaps *caps;
@@ -1224,8 +1327,27 @@ gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
   if (!handled)
     handled = gst_audio_encoder_sink_eventfunc (enc, event);
 
-  if (!handled)
-    ret = gst_pad_event_default (pad, event);
+  if (!handled) {
+    /* Forward non-serialized events and EOS/FLUSH_STOP immediately.
+     * For EOS this is required because no buffer or serialized event
+     * will come after EOS and nothing could trigger another
+     * _finish_frame() call.
+     *
+     * For FLUSH_STOP this is required because it is expected
+     * to be forwarded immediately and no buffers are queued anyway.
+     */
+    if (!GST_EVENT_IS_SERIALIZED (event)
+        || GST_EVENT_TYPE (event) == GST_EVENT_EOS
+        || GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
+      ret = gst_pad_event_default (pad, event);
+    } else {
+      GST_AUDIO_ENCODER_STREAM_LOCK (enc);
+      enc->priv->pending_events =
+          g_list_append (enc->priv->pending_events, event);
+      GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
+      ret = TRUE;
+    }
+  }
 
   GST_DEBUG_OBJECT (enc, "event handled");
 
@@ -1544,6 +1666,11 @@ gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
   GST_DEBUG_OBJECT (enc, "activate %d", active);
 
   if (active) {
+
+    if (enc->priv->tags)
+      gst_tag_list_free (enc->priv->tags);
+    enc->priv->tags = gst_tag_list_new ();
+
     if (!enc->priv->active && klass->start)
       result = klass->start (enc);
   } else {
@@ -1601,37 +1728,77 @@ gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
 }
 
 /**
- * gst_audio_encoder_set_frame_samples:
+ * gst_audio_encoder_set_frame_samples_min:
  * @enc: a #GstAudioEncoder
  * @num: number of samples per frame
  *
  * Sets number of samples (per channel) subclass needs to be handed,
- * or will be handed all available if 0.
+ * at least or will be handed all available if 0.
+ *
+ * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
+ * must be called with the same number.
  *
  * Since: 0.10.36
  */
 void
-gst_audio_encoder_set_frame_samples (GstAudioEncoder * enc, gint num)
+gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num)
 {
   g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
 
-  enc->priv->ctx.frame_samples = num;
+  enc->priv->ctx.frame_samples_min = num;
 }
 
 /**
- * gst_audio_encoder_get_frame_samples:
+ * gst_audio_encoder_get_frame_samples_min:
  * @enc: a #GstAudioEncoder
  *
- * Returns: currently requested samples per frame
+ * Returns: currently minimum requested samples per frame
  *
  * Since: 0.10.36
  */
 gint
-gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc)
+gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc)
 {
   g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
 
-  return enc->priv->ctx.frame_samples;
+  return enc->priv->ctx.frame_samples_min;
+}
+
+/**
+ * gst_audio_encoder_set_frame_samples_max:
+ * @enc: a #GstAudioEncoder
+ * @num: number of samples per frame
+ *
+ * Sets number of samples (per channel) subclass needs to be handed,
+ * at most or will be handed all available if 0.
+ *
+ * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
+ * must be called with the same number.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num)
+{
+  g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
+
+  enc->priv->ctx.frame_samples_max = num;
+}
+
+/**
+ * gst_audio_encoder_get_frame_samples_min:
+ * @enc: a #GstAudioEncoder
+ *
+ * Returns: currently maximum requested samples per frame
+ *
+ * Since: 0.10.36
+ */
+gint
+gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc)
+{
+  g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
+
+  return enc->priv->ctx.frame_samples_max;
 }
 
 /**
@@ -1639,7 +1806,8 @@ gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc)
  * @enc: a #GstAudioEncoder
  * @num: number of frames
  *
- * Sets max number of frames accepted at once (assumed minimally 1)
+ * Sets max number of frames accepted at once (assumed minimally 1).
+ * Requires @frame_samples_min and @frame_samples_max to be the equal.
  *
  * Since: 0.10.36
  */
@@ -1939,3 +2107,40 @@ gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
 
   return result;
 }
+
+/**
+ * gst_audio_encoder_merge_tags:
+ * @enc: a #GstAudioEncoder
+ * @tags: a #GstTagList to merge
+ * @mode: the #GstTagMergeMode to use
+ *
+ * Adds tags to so-called pending tags, which will be processed
+ * before pushing out data downstream.
+ *
+ * Note that this is provided for convenience, and the subclass is
+ * not required to use this and can still do tag handling on its own,
+ * although it should be aware that baseclass already takes care
+ * of the usual CODEC/AUDIO_CODEC tags.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
+    const GstTagList * tags, GstTagMergeMode mode)
+{
+  GstTagList *otags;
+
+  g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
+  g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
+
+  GST_OBJECT_LOCK (enc);
+  if (tags)
+    GST_DEBUG_OBJECT (enc, "merging tags %" GST_PTR_FORMAT, tags);
+  otags = enc->priv->tags;
+  enc->priv->tags = gst_tag_list_merge (enc->priv->tags, tags, mode);
+  if (otags)
+    gst_tag_list_free (otags);
+  GST_OBJECT_UNLOCK (enc);
+}
diff --git a/gst-libs/gst/audio/gstaudioencoder.h b/gst-libs/gst/audio/gstaudioencoder.h
index 1a8c5af06b..7685b18c02 100644
--- a/gst-libs/gst/audio/gstaudioencoder.h
+++ b/gst-libs/gst/audio/gstaudioencoder.h
@@ -87,6 +87,8 @@ G_BEGIN_DECLS
  */
 #define GST_AUDIO_ENCODER_SEGMENT(obj)     (GST_AUDIO_ENCODER_CAST (obj)->segment)
 
+#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
+#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
 
 typedef struct _GstAudioEncoder GstAudioEncoder;
 typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
@@ -108,6 +110,11 @@ struct _GstAudioEncoder {
   GstPad         *sinkpad;
   GstPad         *srcpad;
 
+  /* protects all data processing, i.e. is locked
+   * in the chain function, finish_frame and when
+   * processing serialized events */
+  GStaticRecMutex stream_lock;
+
   /* MT-protected (with STREAM_LOCK) */
   GstSegment      segment;
 
@@ -196,9 +203,13 @@ GstCaps *       gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
 /* context parameters */
 GstAudioInfo  * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
 
-gint            gst_audio_encoder_get_frame_samples (GstAudioEncoder * enc);
+gint            gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
 
-void            gst_audio_encoder_set_frame_samples (GstAudioEncoder * enc, gint num);
+void            gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
+
+gint            gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
+
+void            gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
 
 gint            gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
 
@@ -238,6 +249,9 @@ void            gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
 
 gint64          gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
 
+void            gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
+                                              const GstTagList * tags, GstTagMergeMode mode);
+
 G_END_DECLS
 
 #endif /* __GST_AUDIO_ENCODER_H__ */
diff --git a/gst/typefind/gsttypefindfunctions.c b/gst/typefind/gsttypefindfunctions.c
index 31f516e606..2e2a56702a 100644
--- a/gst/typefind/gsttypefindfunctions.c
+++ b/gst/typefind/gsttypefindfunctions.c
@@ -45,6 +45,10 @@
 GST_DEBUG_CATEGORY_STATIC (type_find_debug);
 #define GST_CAT_DEFAULT type_find_debug
 
+/* so our code stays ready for 0.11 */
+#define gst_type_find_peek(tf,off,len) \
+    ((const guint8 *)gst_type_find_peek((tf),(off),(len)))
+
 /* DataScanCtx: helper for typefind functions that scan through data
  * step-by-step, to avoid doing a peek at each and every offset */
 
diff --git a/win32/common/libgstaudio.def b/win32/common/libgstaudio.def
index cf7e3d43d8..51b3bc6251 100644
--- a/win32/common/libgstaudio.def
+++ b/win32/common/libgstaudio.def
@@ -34,7 +34,8 @@ EXPORTS
 	gst_audio_encoder_finish_frame
 	gst_audio_encoder_get_audio_info
 	gst_audio_encoder_get_frame_max
-	gst_audio_encoder_get_frame_samples
+	gst_audio_encoder_get_frame_samples_max
+	gst_audio_encoder_get_frame_samples_min
 	gst_audio_encoder_get_hard_resync
 	gst_audio_encoder_get_latency
 	gst_audio_encoder_get_lookahead
@@ -42,9 +43,11 @@ EXPORTS
 	gst_audio_encoder_get_perfect_timestamp
 	gst_audio_encoder_get_tolerance
 	gst_audio_encoder_get_type
+	gst_audio_encoder_merge_tags
 	gst_audio_encoder_proxy_getcaps
 	gst_audio_encoder_set_frame_max
-	gst_audio_encoder_set_frame_samples
+	gst_audio_encoder_set_frame_samples_max
+	gst_audio_encoder_set_frame_samples_min
 	gst_audio_encoder_set_hard_resync
 	gst_audio_encoder_set_latency
 	gst_audio_encoder_set_lookahead