eqMerge remote-tracking branch 'origin/master' into 0.11

Conflicts:
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-read-common.c
	gst/rtpmanager/gstrtpssrcdemux.c

13 files changed, 255 insertions, 58 deletions

diff --git a/Android.mk b/Android.mk
index 534fd843c..9ba81d97d 100644
--- a/Android.mk
+++ b/Android.mk
@@ -92,7 +92,7 @@ CONFIGURE_TARGETS += gst-plugins-good-configure
 -include $(GST_PLUGINS_GOOD_TOP)/gst/law/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/level/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/matroska/Android.mk
--include $(GST_PLUGINS_GOOD_TOP)/gst/multifile/Android.mk
+#-include $(GST_PLUGINS_GOOD_TOP)/gst/multifile/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/multipart/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/isomp4/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/replaygain/Android.mk
diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c
index 6138455c9..ad876c2b0 100644
--- a/ext/flac/gstflacenc.c
+++ b/ext/flac/gstflacenc.c
@@ -1047,7 +1047,9 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
       ret = gst_flac_enc_process_stream_headers (flacenc);
       flacenc->got_headers = TRUE;
     }
-  } else if (flacenc->got_headers && samples == 0) {
+  }
+
+  if (flacenc->got_headers && samples == 0) {
     /* header fixup, push downstream directly */
     GST_DEBUG_OBJECT (flacenc, "Fixing up headers at pos=%" G_GUINT64_FORMAT
         ", size=%u", flacenc->offset, (guint) bytes);
diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c
index 8180afbb3..019bef9f8 100644
--- a/ext/jack/gstjack.c
+++ b/ext/jack/gstjack.c
@@ -46,6 +46,27 @@ gst_jack_connect_get_type (void)
   return (GType) jack_connect_type;
 }
 
+GType
+gst_jack_transport_get_type (void)
+{
+  static volatile gsize type = 0;
+
+  if (g_once_init_enter (&type)) {
+    static const GEnumValue enum_values[] = {
+      {GST_JACK_TRANSPORT_AUTONOMOUS,
+          "No transport support", "autonomous"},
+      {GST_JACK_TRANSPORT_MASTER,
+          "Start and stop transport with state changes", "master"},
+      {GST_JACK_TRANSPORT_SLAVE,
+          "Follow transport state changes", "slave"},
+      {0, NULL, NULL},
+    };
+    GType tmp = g_enum_register_static ("GstJackTransport", enum_values);
+    g_once_init_leave (&type, tmp);
+  }
+  return (GType) type;
+}
+
 
 static gpointer
 gst_jack_client_copy (gpointer jclient)
diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h
index 2b5dbe89e..15233ca5c 100644
--- a/ext/jack/gstjack.h
+++ b/ext/jack/gstjack.h
@@ -37,17 +37,32 @@
  *
  * Specify how the output ports will be connected.
  */
-
 typedef enum {
   GST_JACK_CONNECT_NONE,
   GST_JACK_CONNECT_AUTO,
   GST_JACK_CONNECT_AUTO_FORCED
 } GstJackConnect;
 
+/**
+ * GstJackTransport:
+ * @GST_JACK_TRANSPORT_AUTONOMOUS: no transport support
+ * @GST_JACK_TRANSPORT_MASTER: start and stop transport with state-changes
+ * @GST_JACK_TRANSPORT_SLAVE: follow transport state changes
+ *
+ * The jack transport state allow to sync multiple clients. This enum defines a
+ * client behaviour regarding to the transport mechanism.
+ */
+typedef enum {
+  GST_JACK_TRANSPORT_AUTONOMOUS,
+  GST_JACK_TRANSPORT_MASTER,
+  GST_JACK_TRANSPORT_SLAVE
+} GstJackTransport;
+
 typedef jack_default_audio_sample_t sample_t;
 
-#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
-#define GST_TYPE_JACK_CLIENT  (gst_jack_client_get_type ())
+#define GST_TYPE_JACK_CONNECT   (gst_jack_connect_get_type ())
+#define GST_TYPE_JACK_TRANSPORT (gst_jack_transport_get_type ())
+#define GST_TYPE_JACK_CLIENT    (gst_jack_client_get_type ())
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
 #define GST_JACK_FORMAT_STR "F32LE"
@@ -57,5 +72,6 @@ typedef jack_default_audio_sample_t sample_t;
 
 GType gst_jack_client_get_type(void);
 GType gst_jack_connect_get_type(void);
+GType gst_jack_transport_get_type(void);
 
 #endif  // _GST_JACK_H_
diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c
index 2bb355529..c33949f02 100644
--- a/ext/jack/gstjackaudioclient.c
+++ b/ext/jack/gstjackaudioclient.c
@@ -55,6 +55,10 @@ typedef struct
   gint n_clients;
   GList *src_clients;
   GList *sink_clients;
+
+  /* transport state handling */
+  gint cur_ts;
+  GstState transport_state;
 } GstJackAudioConnection;
 
 /* an object sharing a jack_client_t connection. */
@@ -66,7 +70,7 @@ struct _GstJackAudioClient
   gboolean active;
   gboolean deactivate;
 
-  void (*shutdown) (void *arg);
+  JackShutdownCallback shutdown;
   JackProcessCallback process;
   JackBufferSizeCallback buffer_size;
   JackSampleRateCallback sample_rate;
@@ -87,6 +91,27 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
   GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
   GList *walk;
   int res = 0;
+  jack_transport_state_t ts = jack_transport_query (conn->client, NULL);
+
+  if (ts != conn->cur_ts) {
+    conn->cur_ts = ts;
+    switch (ts) {
+      case JackTransportStopped:
+        GST_DEBUG ("transport state is 'stopped'");
+        conn->transport_state = GST_STATE_PAUSED;
+        break;
+      case JackTransportStarting:
+        GST_DEBUG ("transport state is 'starting'");
+        conn->transport_state = GST_STATE_READY;
+        break;
+      case JackTransportRolling:
+        GST_DEBUG ("transport state is 'rolling'");
+        conn->transport_state = GST_STATE_PLAYING;
+        break;
+      default:
+        break;
+    }
+  }
 
   g_mutex_lock (conn->lock);
   /* call sources first, then sinks. Sources will either push data into the
@@ -117,7 +142,6 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
     }
   }
   g_mutex_unlock (conn->lock);
-
   return res;
 }
 
@@ -224,6 +248,8 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server,
   conn->n_clients = 0;
   conn->src_clients = NULL;
   conn->sink_clients = NULL;
+  conn->cur_ts = -1;
+  conn->transport_state = GST_STATE_VOID_PENDING;
 
   /* set our callbacks  */
   jack_set_process_callback (jclient, jack_process_cb, conn);
@@ -525,3 +551,22 @@ gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
 
   return 0;
 }
+
+/**
+ * gst_jack_audio_client_get_transport_state:
+ * @client: a #GstJackAudioClient
+ *
+ * Check the current transport state. The client can use this to request a state
+ * change from the application.
+ *
+ * Returns: the state, %GST_STATE_VOID_PENDING for no change in the transport
+ * state
+ */
+GstState
+gst_jack_audio_client_get_transport_state (GstJackAudioClient * client)
+{
+  GstState state = client->conn->transport_state;
+
+  client->conn->transport_state = GST_STATE_VOID_PENDING;
+  return state;
+}
diff --git a/ext/jack/gstjackaudioclient.h b/ext/jack/gstjackaudioclient.h
index 5fb7e3544..f2669c9f8 100644
--- a/ext/jack/gstjackaudioclient.h
+++ b/ext/jack/gstjackaudioclient.h
@@ -54,6 +54,8 @@ jack_client_t *       gst_jack_audio_client_get_client     (GstJackAudioClient *
 
 gboolean              gst_jack_audio_client_set_active     (GstJackAudioClient *client, gboolean active);
 
+GstState              gst_jack_audio_client_get_transport_state (GstJackAudioClient *client);
+
 G_END_DECLS
 
 #endif /* __GST_JACK_AUDIO_CLIENT_H__ */
diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
index 8055f9219..8bdb664ce 100644
--- a/ext/jack/gstjackaudiosink.c
+++ b/ext/jack/gstjackaudiosink.c
@@ -197,6 +197,18 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
 
   channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
 
+  /* handle transport state requisitions */
+  if (sink->transport == GST_JACK_TRANSPORT_SLAVE) {
+    GstState state = gst_jack_audio_client_get_transport_state (sink->client);
+
+    if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (sink) != state)) {
+      GST_DEBUG_OBJECT (sink, "requesting state change: %s",
+          gst_element_state_get_name (state));
+      gst_element_post_message (GST_ELEMENT (sink),
+          gst_message_new_request_state (GST_OBJECT (sink), state));
+    }
+  }
+
   /* get target buffers */
   for (i = 0; i < channels; i++) {
     sink->buffers[i] =
@@ -564,6 +576,13 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (sink, "start");
 
+  if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_start (client);
+  }
+
   return TRUE;
 }
 
@@ -576,6 +595,13 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (sink, "pause");
 
+  if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_stop (client);
+  }
+
   return TRUE;
 }
 
@@ -588,6 +614,13 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (sink, "stop");
 
+  if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_stop (client);
+  }
+
   return TRUE;
 }
 
@@ -654,7 +687,8 @@ enum
 
 #define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
 #define DEFAULT_PROP_SERVER 		NULL
-#define DEFAULT_PROP_CLIENT_NAME 	NULL
+#define DEFAULT_PROP_CLIENT_NAME	NULL
+#define DEFAULT_PROP_TRANSPORT	GST_JACK_TRANSPORT_AUTONOMOUS
 
 enum
 {
@@ -663,6 +697,7 @@ enum
   PROP_SERVER,
   PROP_CLIENT,
   PROP_CLIENT_NAME,
+  PROP_TRANSPORT,
   PROP_LAST
 };
 
@@ -730,6 +765,19 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
           GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
           G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstJackAudioSink:transport
+   *
+   * The jack transport behaviour for the client.
+   *
+   * Since: 0.10.31
+   */
+  g_object_class_install_property (gobject_class, PROP_TRANSPORT,
+      g_param_spec_enum ("transport", "Transport mode",
+          "Jack transport behaviour of the client",
+          GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   gst_element_class_set_details_simple (gstelement_class, "Audio Sink (Jack)",
       "Sink/Audio", "Output audio to a JACK server",
       "Wim Taymans <wim.taymans@gmail.com>");
@@ -757,8 +805,9 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink)
   sink->jclient = NULL;
   sink->ports = NULL;
   sink->port_count = 0;
-  sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
   sink->buffers = NULL;
+  sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
+  sink->transport = DEFAULT_PROP_TRANSPORT;
 }
 
 static void
@@ -802,6 +851,9 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
         sink->jclient = g_value_get_boxed (value);
       }
       break;
+    case PROP_TRANSPORT:
+      sink->transport = g_value_get_enum (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -829,6 +881,9 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
     case PROP_CLIENT:
       g_value_set_boxed (value, sink->jclient);
       break;
+    case PROP_TRANSPORT:
+      g_value_set_enum (value, sink->transport);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h
index 0dbc142e0..207328676 100644
--- a/ext/jack/gstjackaudiosink.h
+++ b/ext/jack/gstjackaudiosink.h
@@ -59,6 +59,7 @@ struct _GstJackAudioSink {
   gchar           *server;
   jack_client_t   *jclient;
   gchar           *client_name;
+  GstJackTransport transport;
 
   /* our client */
   GstJackAudioClient *client;
diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c
index cf21cbfc1..417028c13 100644
--- a/ext/jack/gstjackaudiosrc.c
+++ b/ext/jack/gstjackaudiosrc.c
@@ -217,6 +217,18 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
 
   channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
 
+  /* handle transport state requisitions */
+  if (src->transport == GST_JACK_TRANSPORT_SLAVE) {
+    GstState state = gst_jack_audio_client_get_transport_state (src->client);
+
+    if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (src) != state)) {
+      GST_DEBUG_OBJECT (src, "requesting state change: %s",
+          gst_element_state_get_name (state));
+      gst_element_post_message (GST_ELEMENT (src),
+          gst_message_new_request_state (GST_OBJECT (src), state));
+    }
+  }
+
   /* get input buffers */
   for (i = 0; i < channels; i++)
     src->buffers[i] =
@@ -576,6 +588,13 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (src, "start");
 
+  if (src->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_start (client);
+  }
+
   return TRUE;
 }
 
@@ -588,6 +607,13 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (src, "pause");
 
+  if (src->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_stop (client);
+  }
+
   return TRUE;
 }
 
@@ -600,6 +626,13 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
 
   GST_DEBUG_OBJECT (src, "stop");
 
+  if (src->transport == GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_stop (client);
+  }
+
   return TRUE;
 }
 
@@ -658,6 +691,7 @@ enum
 #define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
 #define DEFAULT_PROP_SERVER 		NULL
 #define DEFAULT_PROP_CLIENT_NAME	NULL
+#define DEFAULT_PROP_TRANSPORT	GST_JACK_TRANSPORT_AUTONOMOUS
 
 enum
 {
@@ -666,6 +700,7 @@ enum
   PROP_SERVER,
   PROP_CLIENT,
   PROP_CLIENT_NAME,
+  PROP_TRANSPORT,
   PROP_LAST
 };
 
@@ -751,6 +786,19 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
           GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
           G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstJackAudioSink:transport
+   *
+   * The jack transport behaviour for the client.
+   *
+   * Since: 0.10.31
+   */
+  g_object_class_install_property (gobject_class, PROP_TRANSPORT,
+      g_param_spec_enum ("transport", "Transport mode",
+          "Jack transport behaviour of the client",
+          GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&src_factory));
 
@@ -769,11 +817,6 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
   gst_jack_audio_client_init ();
 }
 
-/* initialize the new element
- * instantiate pads and add them to element
- * set pad calback functions
- * initialize instance structure
- */
 static void
 gst_jack_audio_src_init (GstJackAudioSrc * src)
 {
@@ -785,6 +828,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src)
   src->port_count = 0;
   src->buffers = NULL;
   src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
+  src->transport = DEFAULT_PROP_TRANSPORT;
 }
 
 static void
@@ -826,6 +870,9 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id,
         src->jclient = g_value_get_boxed (value);
       }
       break;
+    case PROP_TRANSPORT:
+      src->transport = g_value_get_enum (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -851,6 +898,9 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id,
     case PROP_CLIENT:
       g_value_set_boxed (value, src->jclient);
       break;
+    case PROP_TRANSPORT:
+      g_value_set_enum (value, src->transport);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h
index 1401d9b0d..ed8f59333 100644
--- a/ext/jack/gstjackaudiosrc.h
+++ b/ext/jack/gstjackaudiosrc.h
@@ -76,6 +76,7 @@ struct _GstJackAudioSrc
     gchar           *server;
     jack_client_t   *jclient;
     gchar           *client_name;
+    GstJackTransport transport;
 
     /* our client */
     GstJackAudioClient *client;
diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c
index 5fe25f957..442f64ef9 100644
--- a/gst/matroska/matroska-mux.c
+++ b/gst/matroska/matroska-mux.c
@@ -668,6 +668,17 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstObject * parent,
   return gst_pad_event_default (pad, parent, event);
 }
 
+
+static void
+gst_matroska_mux_free_codec_priv (GstMatroskaTrackContext * context)
+{
+  if (context->codec_priv != NULL) {
+    g_free (context->codec_priv);
+    context->codec_priv = NULL;
+    context->codec_priv_size = 0;
+  }
+}
+
 static void
 gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context,
     const guint * clut)
@@ -695,7 +706,7 @@ gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context,
   sclut = g_strjoinv (",", clutv);
 
   /* build codec private; only palette for now */
-  g_free (context->codec_priv);
+  gst_matroska_mux_free_codec_priv (context);
   context->codec_priv = (guint8 *) g_strdup_printf ("palette: %s", sclut);
   /* include terminating 0 */
   context->codec_priv_size = strlen ((gchar *) context->codec_priv) + 1;
@@ -1046,18 +1057,13 @@ skip_details:
 
     gst_matroska_mux_set_codec_id (context,
         GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC);
+    gst_matroska_mux_free_codec_priv (context);
     context->codec_priv = (gpointer) bih;
     context->codec_priv_size = size;
   } else if (!strcmp (mimetype, "video/x-h264")) {
     gst_matroska_mux_set_codec_id (context,
         GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC);
-
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
-
+    gst_matroska_mux_free_codec_priv (context);
     /* Create avcC header */
     if (codec_buf != NULL) {
       context->codec_priv_size = gst_buffer_get_size (codec_buf);
@@ -1069,11 +1075,7 @@ skip_details:
 
     gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_THEORA);
 
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
+    gst_matroska_mux_free_codec_priv (context);
 
     streamheader = gst_structure_get_value (structure, "streamheader");
     if (!theora_streamheader_to_codecdata (streamheader, context)) {
@@ -1108,6 +1110,7 @@ skip_details:
 
     /* global headers may be in codec data */
     if (codec_buf != NULL) {
+      gst_matroska_mux_free_codec_priv (context);
       context->codec_priv_size = gst_buffer_get_size (codec_buf);
       context->codec_priv = g_malloc0 (context->codec_priv_size);
       gst_buffer_extract (codec_buf, 0, context->codec_priv, -1);
@@ -1154,6 +1157,7 @@ skip_details:
 
       gst_buffer_extract (codec_data_buf, 0, priv_data, -1);
 
+      gst_matroska_mux_free_codec_priv (context);
       context->codec_priv = priv_data;
       context->codec_priv_size = priv_data_size;
     }
@@ -1237,6 +1241,7 @@ xiphN_streamheader_to_codecdata (const GValue * streamheader,
     offset += gst_buffer_get_size (buf[i]);
   }
 
+  gst_matroska_mux_free_codec_priv (context);
   context->codec_priv = priv_data;
   context->codec_priv_size = priv_data_size;
 
@@ -1421,6 +1426,7 @@ flac_streamheader_to_codecdata (const GValue * streamheader,
     return FALSE;
   }
 
+  gst_matroska_mux_free_codec_priv (context);
   context->codec_priv_size = gst_buffer_get_size (buffer) - 9;
   context->codec_priv = g_malloc (context->codec_priv_size);
   gst_buffer_extract (buffer, 9, context->codec_priv, -1);
@@ -1430,9 +1436,7 @@ flac_streamheader_to_codecdata (const GValue * streamheader,
     bufval = &g_array_index (bufarr, GValue, i);
 
     if (G_VALUE_TYPE (bufval) != GST_TYPE_BUFFER) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
+      gst_matroska_mux_free_codec_priv (context);
       GST_WARNING ("streamheaders array does not contain GstBuffers");
       return FALSE;
     }
@@ -1487,6 +1491,7 @@ speex_streamheader_to_codecdata (const GValue * streamheader,
     return FALSE;
   }
 
+  gst_matroska_mux_free_codec_priv (context);
   context->codec_priv_size = gst_buffer_get_size (buffer);
   context->codec_priv = g_malloc (context->codec_priv_size);
   gst_buffer_extract (buffer, 0, context->codec_priv, -1);
@@ -1494,9 +1499,7 @@ speex_streamheader_to_codecdata (const GValue * streamheader,
   bufval = &g_array_index (bufarr, GValue, 1);
 
   if (G_VALUE_TYPE (bufval) != GST_TYPE_BUFFER) {
-    g_free (context->codec_priv);
-    context->codec_priv = NULL;
-    context->codec_priv_size = 0;
+    gst_matroska_mux_free_codec_priv (context);
     GST_WARNING ("streamheaders array does not contain GstBuffers");
     return FALSE;
   }
@@ -1729,11 +1732,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
 
     gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_VORBIS);
 
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
+    gst_matroska_mux_free_codec_priv (context);
 
     streamheader = gst_structure_get_value (structure, "streamheader");
     if (!vorbis_streamheader_to_codecdata (streamheader, context)) {
@@ -1745,11 +1744,8 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
     const GValue *streamheader;
 
     gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_FLAC);
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
+
+    gst_matroska_mux_free_codec_priv (context);
 
     streamheader = gst_structure_get_value (structure, "streamheader");
     if (!flac_streamheader_to_codecdata (streamheader, context)) {
@@ -1761,11 +1757,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
     const GValue *streamheader;
 
     gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_SPEEX);
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
+    gst_matroska_mux_free_codec_priv (context);
 
     streamheader = gst_structure_get_value (structure, "streamheader");
     if (!speex_streamheader_to_codecdata (streamheader, context)) {
@@ -1823,6 +1815,8 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
 
       gst_buffer_extract (codec_data_buf, 0, priv_data, -1);
 
+      gst_matroska_mux_free_codec_priv (context);
+
       context->codec_priv = priv_data;
       context->codec_priv_size = priv_data_size;
     }
@@ -1907,6 +1901,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps)
     }
 
     gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_ACM);
+    gst_matroska_mux_free_codec_priv (context);
     context->codec_priv = (gpointer) codec_priv;
     context->codec_priv_size = codec_priv_size;
   }
@@ -1981,11 +1976,7 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
     gst_matroska_mux_set_codec_id (context,
         GST_MATROSKA_CODEC_ID_SUBTITLE_KATE);
 
-    if (context->codec_priv != NULL) {
-      g_free (context->codec_priv);
-      context->codec_priv = NULL;
-      context->codec_priv_size = 0;
-    }
+    gst_matroska_mux_free_codec_priv (context);
 
     streamheader = gst_structure_get_value (structure, "streamheader");
     if (!kate_streamheader_to_codecdata (streamheader, context)) {
@@ -2030,8 +2021,7 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
       return TRUE;
     }
 
-    if (context->codec_priv != NULL)
-      g_free (context->codec_priv);
+    gst_matroska_mux_free_codec_priv (context);
 
     priv_data = g_malloc0 (priv_data_size);
     memcpy (priv_data, priv_buffer_data, priv_data_size);
diff --git a/gst/matroska/matroska-read-common.c b/gst/matroska/matroska-read-common.c
index 0b45855df..2f147eaaa 100644
--- a/gst/matroska/matroska-read-common.c
+++ b/gst/matroska/matroska-read-common.c
@@ -650,6 +650,9 @@ gst_matroska_read_common_parse_attached_file (GstMatroskaReadCommon * common,
     else
       gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_ATTACHMENT,
           tagsample, NULL);
+
+    /* the list adds it own ref */
+    gst_buffer_unref (tagsample);
   }
 
   g_free (filename);
diff --git a/tests/check/pipelines/wavenc.c b/tests/check/pipelines/wavenc.c
index ab9f62b41..c5344a75d 100644
--- a/tests/check/pipelines/wavenc.c
+++ b/tests/check/pipelines/wavenc.c
@@ -54,7 +54,7 @@ bus_handler (GstBus * bus, GstMessage * message, gpointer data)
 
 /*
  * gst-launch \
- * audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! wavenc ! filesink location=/tmp/mc.wav \
+ * audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! audioconvert ! wavenc ! filesink location=/tmp/mc.wav \
  * audiotestsrc freq=880 num-buffers=100 ! i.
  * ...
  *
@@ -65,7 +65,7 @@ static void
 make_n_channel_wav (const gint channels, const GValueArray * arr)
 {
   GstElement *pipeline;
-  GstElement **audiotestsrc, *interleave, *wavenc, *fakesink;
+  GstElement **audiotestsrc, *interleave, *wavenc, *conv, *fakesink;
   GstBus *bus;
   GMainLoop *loop;
   guint i;
@@ -81,10 +81,21 @@ make_n_channel_wav (const gint channels, const GValueArray * arr)
   g_object_set (interleave, "channel-positions", arr, NULL);
   gst_bin_add (GST_BIN (pipeline), interleave);
 
+  if (G_BYTE_ORDER == G_BIG_ENDIAN) {
+    /* we're not here to test orc; audioconvert misbehaves on ppc32 */
+    g_setenv ("ORC_CODE", "backup", 1);
+    conv = gst_element_factory_make ("audioconvert", NULL);
+  } else {
+    conv = gst_element_factory_make ("identity", NULL);
+  }
+  fail_unless (conv != NULL);
+  gst_bin_add (GST_BIN (pipeline), conv);
+  fail_unless (gst_element_link (interleave, conv));
+
   wavenc = gst_element_factory_make ("wavenc", NULL);
   fail_unless (wavenc != NULL);
   gst_bin_add (GST_BIN (pipeline), wavenc);
-  fail_unless (gst_element_link (interleave, wavenc));
+  fail_unless (gst_element_link (conv, wavenc));
 
   fakesink = gst_element_factory_make ("fakesink", NULL);
   fail_unless (fakesink != NULL);