Merge branch 'master' into 0.11

Conflicts:
	ext/vorbis/gstvorbisdec.c
	ext/vorbis/gstvorbisenc.c
	ext/vorbis/gstvorbisenc.h
	gst/audiotestsrc/gstaudiotestsrc.c

ext/vorbis/Makefile.am

@@ -11,7 +11,8 @@ libgstvorbis_la_SOURCES = gstvorbis.c \
 			  gstvorbistag.c \
 			  gstvorbiscommon.c
 
-libgstvorbis_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VORBIS_CFLAGS)
+libgstvorbis_la_CFLAGS = -DGST_USE_UNSTABLE_API \
+	$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(VORBIS_CFLAGS)
 ## AM_PATH_VORBIS also sets VORBISENC_LIBS
 libgstvorbis_la_LIBADD = \
 	$(top_builddir)/gst-libs/gst/tag/libgsttag-@GST_MAJORMINOR@.la \
@@ -27,7 +28,8 @@ plugin_LTLIBRARIES += libgstivorbisdec.la
 
 libgstivorbisdec_la_SOURCES = gstivorbisdec.c \
 	gstvorbisdec.c gstvorbisdeclib.c gstvorbiscommon.c
-libgstivorbisdec_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
+libgstivorbisdec_la_CFLAGS = -DGST_USE_UNSTABLE_API \
+	$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
 	-DTREMOR $(IVORBIS_CFLAGS)
 libgstivorbisdec_la_LIBADD = \
 	$(top_builddir)/gst-libs/gst/tag/libgsttag-@GST_MAJORMINOR@.la \

ext/vorbis/gstvorbisdec.h

@@ -27,6 +27,7 @@
 #endif
 
 #include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
 #include "gstvorbisdeclib.h"
 
 G_BEGIN_DECLS
@@ -51,15 +52,11 @@ typedef struct _GstVorbisDecClass GstVorbisDecClass;
  * Opaque data structure.
  */
 struct _GstVorbisDec {
-  GstElement        element;
+  GstAudioDecoder element;
-
-  GstPad           *sinkpad;
-  GstPad           *srcpad;
 
   vorbis_dsp_state  vd;
   vorbis_info       vi;
   vorbis_comment    vc;
-
 #ifndef USE_TREMOLO
   vorbis_block      vb;
 #endif
@@ -67,26 +64,13 @@ struct _GstVorbisDec {
   gboolean          initialized;
   GstAudioInfo      info;
 
-  /* list of buffers that need timestamps */
-  GList            *queued;
-  /* gather/decode queues for reverse playback */
-  GList            *gather;
-  GList            *decode;
-
-  GstSegment        segment;
-  gboolean          discont;
-  guint32           seqnum;
-
-  GstClockTime      last_timestamp;
-
-  GList            *pendingevents;
   GstTagList       *taglist;
 
   CopySampleFunc    copy_samples;
 };
 
 struct _GstVorbisDecClass {
-  GstElementClass parent_class;
+  GstAudioDecoderClass parent_class;
 };
 
 GType gst_vorbis_dec_get_type(void);

ext/vorbis/gstvorbisenc.h

@@ -23,6 +23,7 @@
 
 
 #include <gst/gst.h>
+#include <gst/audio/gstaudioencoder.h>
 
 #include <vorbis/codec.h>
 
@@ -48,14 +49,11 @@ typedef struct _GstVorbisEncClass GstVorbisEncClass;
  * Opaque data structure.
  */
 struct _GstVorbisEnc {
-  GstElement       element;
+  GstAudioEncoder element;
 
-  GstPad          *sinkpad;
-  GstPad          *srcpad;
-
-  GstCaps         *srccaps;
   GstCaps         *sinkcaps;
 
+  /* codec */
   vorbis_info      vi; /* struct that stores all the static vorbis bitstream
                                                             settings */
   vorbis_comment   vc; /* struct that stores all the user comments */
@@ -63,6 +61,7 @@ struct _GstVorbisEnc {
   vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
   vorbis_block     vb; /* local working space for packet->PCM decode */
 
+  /* properties */
   gboolean         managed;
   gint             bitrate;
   gint             min_bitrate;
@@ -74,13 +73,8 @@ struct _GstVorbisEnc {
   gint             frequency;
 
   guint64          samples_in;
+  guint64          samples_out;
   guint64          bytes_out;
-  GstClockTime     next_ts;
-  GstClockTime     expected_ts;
-  gboolean         next_discont;
-  guint64          granulepos_offset;
-  gint64           subgranule_offset;
-  GstSegment       segment;
 
   GstTagList *     tags;
 
@@ -90,7 +84,7 @@ struct _GstVorbisEnc {
 };
 
 struct _GstVorbisEncClass {
-  GstElementClass parent_class;
+  GstAudioEncoderClass parent_class;
 };
 
 GType gst_vorbis_enc_get_type(void);

gst-libs/gst/audio/gstaudiodecoder.c

@@ -1579,6 +1579,10 @@ gst_audio_decoder_src_event (GstPad * pad, GstEvent * event)
   gboolean res = FALSE;
 
   dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
+  if (G_UNLIKELY (dec == NULL)) {
+    gst_event_unref (event);
+    return FALSE;
+  }
 
   GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
       GST_EVENT_TYPE_NAME (event));
@@ -1779,6 +1783,9 @@ gst_audio_decoder_src_query (GstPad * pad, GstQuery * query)
   gboolean res = FALSE;
 
   dec = GST_AUDIO_DECODER (GST_PAD_PARENT (pad));
+  if (G_UNLIKELY (dec == NULL))
+    return FALSE;
+
   peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
 
   GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);

gst-libs/gst/audio/gstaudioencoder.c

@@ -1504,6 +1504,9 @@ gst_audio_encoder_src_query (GstPad * pad, GstQuery * query)
   gboolean res = FALSE;
 
   enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
+  if (G_UNLIKELY (enc == NULL))
+    return FALSE;
+
   peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
 
   GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);

gst/audiotestsrc/gstaudiotestsrc.c

@@ -789,7 +789,7 @@ gst_audio_test_src_create_red_noise_##type (GstAudioTestSrc * src, g##type * sam
       while (TRUE) { \
         gdouble r = g_rand_double_range (src->gen, -1.0, 1.0); \
         state += r; \
-        if (state<-8.0f || state>8.0f) state -= r; \
+        if (state < -8.0f || state > 8.0f) state -= r; \
         else break; \
       } \
       samples[i++] = (g##type) (amp * state * 0.0625f); /* /16.0 */ \
@@ -958,6 +958,7 @@ gst_audio_test_src_change_wave (GstAudioTestSrc * src)
         src->gen = g_rand_new ();
       src->red.state = 0.0;
       src->process = violet_noise_funcs[idx];
+      break;
     default:
       GST_ERROR ("invalid wave-form");
       break;

gst/encoding/gstencodebin.c

@@ -1090,45 +1090,6 @@ _create_stream_group (GstEncodeBin * ebin, GstEncodingProfile * sprof,
   last = sgroup->outfilter;
 
 
-  /* FIXME :
-   *
-   *   The usage of parsers in encoding/muxing scenarios is
-   * just too undefined to just use as-is.
-   *
-   * Take the use-case where you want to re-mux a stream of type
-   * "my/media". You create a StreamEncodingProfile with that type
-   * as the target (as-is). And you use decodebin2/uridecodebin
-   * upstream.
-   *
-   * * demuxer exposes "my/media"
-   * * a parser is available for "my/media" which has a source pad
-   *   caps of "my/media,parsed=True"
-   * * decodebin2/uridecodebin exposes a new pad with the parsed caps
-   * * You request a new stream from encodebin, which will match the
-   *   streamprofile and creates a group (i.e. going through this method)
-   *   There is a matching parser (the same used in the decoder) whose
-   *   source pad caps intersects with the stream profile caps, you
-   *   therefore use it...
-   * * ... but that parser has a "my/media,parsed=False" sink pad caps
-   * * ... and you can't link your decodebin pad to encodebin.
-   *
-   * In the end, it comes down to parsers only taking into account the
-   * decoding use-cases.
-   *
-   * One way to solve that might be to :
-   * * Make parsers sink pad caps be "framed={False,True}" and the
-   *   source pad caps be "framed=True"
-   * * Modify decodebin2 accordingly to avoid looping and chaining
-   *   an infinite number of parsers
-   *
-   * Another way would be to have "well-known" caps properties to specify
-   * whether a stream has been parsed or not.
-   * * currently we fail. aacparse uses 'framed' and mp3parse uses 'parsed'
-   */
-  /* FIXME : Re-enable once parser situation is un-$#*@(%$#ed */
-#if 0
-  /* Parser.
-   * FIXME : identify smart parsers (used for re-encoding) */
   sgroup->parser = _get_parser (ebin, sprof);
 
   if (sgroup->parser != NULL) {
@@ -1139,7 +1100,6 @@ _create_stream_group (GstEncodeBin * ebin, GstEncodingProfile * sprof,
       goto parser_link_failure;
     last = sgroup->parser;
   }
-#endif
 
   /* Stream combiner */
   sgroup->combiner = g_object_new (GST_TYPE_STREAM_COMBINER, NULL);
@@ -1467,11 +1427,9 @@ combiner_link_failure:
   GST_ERROR_OBJECT (ebin, "Failure linking to the combiner");
   goto cleanup;
 
-#if 0
 parser_link_failure:
   GST_ERROR_OBJECT (ebin, "Failure linking the parser");
   goto cleanup;
-#endif
 
 converter_link_failure:
   GST_ERROR_OBJECT (ebin, "Failure linking the video converters");

gst/playback/gstplaybin2.c

@@ -2445,12 +2445,14 @@ stream_changed_data_probe (GstPad * pad, GstProbeType type,
 
 /* helper function to lookup stuff in lists */
 static gboolean
-array_has_value (const gchar * values[], const gchar * value)
+array_has_value (const gchar * values[], const gchar * value, gboolean exact)
 {
   gint i;
 
   for (i = 0; values[i]; i++) {
-    if (values[i] && g_str_has_prefix (value, values[i]))
+    if (exact && !strcmp (value, values[i]))
+      return TRUE;
+    if (!exact && g_str_has_prefix (value, values[i]))
       return TRUE;
   }
   return FALSE;
@@ -2508,7 +2510,7 @@ pad_added_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
   GstPad *sinkpad;
   GstPadLinkReturn res;
   GstSourceSelect *select = NULL;
-  gint i;
+  gint i, pass;
   gboolean changed = FALSE;
 
   playbin = group->playbin;
@@ -2521,9 +2523,12 @@ pad_added_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
       "pad %s:%s with caps %" GST_PTR_FORMAT " added in group %p",
       GST_DEBUG_PAD_NAME (pad), caps, group);
 
-  /* major type of the pad, this determines the selector to use */
+  /* major type of the pad, this determines the selector to use,
+     try exact match first so we don't prematurely match video/
+     for video/x-dvd-subpicture */
+  for (pass = 0; !select && pass < 2; pass++) {
     for (i = 0; i < PLAYBIN_STREAM_LAST; i++) {
-    if (array_has_value (group->selector[i].media_list, name)) {
+      if (array_has_value (group->selector[i].media_list, name, pass == 0)) {
         select = &group->selector[i];
         break;
       } else if (group->selector[i].get_media_caps) {
@@ -2537,6 +2542,7 @@ pad_added_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
         gst_caps_unref (media_caps);
       }
     }
+  }
   /* no selector found for the media type, don't bother linking it to a
    * selector. This will leave the pad unlinked and thus ignored. */
   if (select == NULL)

gst/playback/gstplaysink.c

@@ -255,6 +255,8 @@ enum
   PROP_FRAME,
   PROP_AV_OFFSET,
   PROP_VIDEO_SINK,
+  PROP_AUDIO_SINK,
+  PROP_TEXT_SINK,
   PROP_LAST
 };
 
@@ -422,6 +424,31 @@ gst_play_sink_class_init (GstPlaySinkClass * klass)
       g_param_spec_object ("video-sink", "Video Sink",
           "the video output element to use (NULL = default sink)",
           GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstPlaySink:audio-sink:
+   *
+   * Set the used audio sink element. NULL will use the default sink. playsink
+   * must be in %GST_STATE_NULL
+   *
+   * Since: 0.10.36
+   */
+  g_object_class_install_property (gobject_klass, PROP_AUDIO_SINK,
+      g_param_spec_object ("audio-sink", "Audio Sink",
+          "the audio output element to use (NULL = default sink)",
+          GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstPlaySink:text-sink:
+   *
+   * Set the used text sink element. NULL will use the default sink. playsink
+   * must be in %GST_STATE_NULL
+   *
+   * Since: 0.10.36
+   */
+  g_object_class_install_property (gobject_klass, PROP_TEXT_SINK,
+      g_param_spec_object ("text-sink", "Text sink",
+          "the text output element to use (NULL = default textoverlay)",
+          GST_TYPE_ELEMENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
 
   g_signal_new ("reconfigure", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstPlaySinkClass,
@@ -3628,6 +3655,14 @@ gst_play_sink_set_property (GObject * object, guint prop_id,
       gst_play_sink_set_sink (playsink, GST_PLAY_SINK_TYPE_VIDEO,
           g_value_get_object (value));
       break;
+    case PROP_AUDIO_SINK:
+      gst_play_sink_set_sink (playsink, GST_PLAY_SINK_TYPE_AUDIO,
+          g_value_get_object (value));
+      break;
+    case PROP_TEXT_SINK:
+      gst_play_sink_set_sink (playsink, GST_PLAY_SINK_TYPE_TEXT,
+          g_value_get_object (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, spec);
       break;
@@ -3670,6 +3705,14 @@ gst_play_sink_get_property (GObject * object, guint prop_id,
       g_value_take_object (value, gst_play_sink_get_sink (playsink,
               GST_PLAY_SINK_TYPE_VIDEO));
       break;
+    case PROP_AUDIO_SINK:
+      g_value_take_object (value, gst_play_sink_get_sink (playsink,
+              GST_PLAY_SINK_TYPE_AUDIO));
+      break;
+    case PROP_TEXT_SINK:
+      g_value_take_object (value, gst_play_sink_get_sink (playsink,
+              GST_PLAY_SINK_TYPE_TEXT));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, spec);
       break;

gst/playback/gstplaysinkaudioconvert.c

@@ -339,10 +339,7 @@ gst_play_sink_audio_convert_getcaps (GstPad * pad, GstCaps * filter)
   GstPad *otherpad, *peer;
 
   GST_PLAY_SINK_AUDIO_CONVERT_LOCK (self);
-  if (pad == self->srcpad)
+  otherpad = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
-    otherpad = gst_object_ref (self->sinkpad);
-  else
-    otherpad = gst_object_ref (self->srcpad);
   GST_PLAY_SINK_AUDIO_CONVERT_UNLOCK (self);
 
   peer = gst_pad_get_peer (otherpad);

gst/playback/gstplaysinkvideoconvert.c

@@ -319,10 +319,7 @@ gst_play_sink_video_convert_getcaps (GstPad * pad, GstCaps * filter)
   GstPad *otherpad, *peer;
 
   GST_PLAY_SINK_VIDEO_CONVERT_LOCK (self);
-  if (pad == self->srcpad)
+  otherpad = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
-    otherpad = gst_object_ref (self->sinkpad);
-  else
-    otherpad = gst_object_ref (self->srcpad);
   GST_PLAY_SINK_VIDEO_CONVERT_UNLOCK (self);
 
   peer = gst_pad_get_peer (otherpad);

tests/check/elements/audiotestsrc.c

@@ -87,6 +87,7 @@ GST_START_TEST (test_all_waves)
 
   while (values[j].value_name) {
     GST_DEBUG_OBJECT (audiotestsrc, "testing wave %s", values[j].value_name);
+    g_object_set (audiotestsrc, "wave", values[j].value, NULL);
 
     fail_unless (gst_element_set_state (audiotestsrc,
             GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,