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

diff --cc ext/flac/gstflacenc.c
Simple merge

diff --cc ext/jack/gstjack.h
index 2b5dbe8,d00c0e7..15233ca
--- 1/ext/jack/gstjack.h
--- 2/ext/jack/gstjack.h
+++ b/ext/jack/gstjack.h
@@@ -44,18 -43,29 +43,35 @@@ typedef enum 
    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"
 +#else
 +#define GST_JACK_FORMAT_STR "F32BE"
 +#endif
 +
  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 --cc ext/jack/gstjackaudiosink.c
index 8055f92,681671a..8bdb664
--- 1/ext/jack/gstjackaudiosink.c
--- 2/ext/jack/gstjackaudiosink.c
+++ b/ext/jack/gstjackaudiosink.c
@@@ -192,11 -192,23 +192,23 @@@ jack_process_cb (jack_nframes_t nframes
    gint i, j, flen, channels;
    sample_t *data;
  
 -  buf = GST_RING_BUFFER_CAST (arg);
 +  buf = GST_AUDIO_RING_BUFFER_CAST (arg);
    sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
  
 -  channels = buf->spec.channels;
 -
 +  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] =
@@@ -730,16 -771,22 +765,29 @@@ gst_jack_audio_sink_class_init (GstJack
            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>");
 +
 +  gst_element_class_add_pad_template (gstelement_class,
 +      gst_static_pad_template_get (&jackaudiosink_sink_factory));
 +
    gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
  
 -  gstbaseaudiosink_class->create_ringbuffer =
 +  gstaudiobasesink_class->create_ringbuffer =
        GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
  
    /* ref class from a thread-safe context to work around missing bit of

diff --cc ext/jack/gstjackaudiosink.h
Simple merge

diff --cc ext/jack/gstjackaudiosrc.c
index cf21cbf,b85fe40..417028c
--- 1/ext/jack/gstjackaudiosrc.c
--- 2/ext/jack/gstjackaudiosrc.c
+++ b/ext/jack/gstjackaudiosrc.c
@@@ -212,11 -212,23 +212,23 @@@ jack_process_cb (jack_nframes_t nframes
    gint channels, i, j, flen;
    sample_t *data;
  
 -  buf = GST_RING_BUFFER_CAST (arg);
 +  buf = GST_AUDIO_RING_BUFFER_CAST (arg);
    src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
  
 -  channels = buf->spec.channels;
 -
 +  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] =
@@@ -751,15 -791,21 +786,28 @@@ gst_jack_audio_src_class_init (GstJackA
            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));
 +
 +  gst_element_class_set_details_simple (gstelement_class, "Audio Source (Jack)",
 +      "Source/Audio", "Captures audio from a JACK server",
 +      "Tristan Matthews <tristan@sat.qc.ca>");
 +
    gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
 -  gstbaseaudiosrc_class->create_ringbuffer =
 +  gstaudiobasesrc_class->create_ringbuffer =
        GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
  
    /* ref class from a thread-safe context to work around missing bit of
@@@ -769,13 -815,8 +817,8 @@@
    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, GstJackAudioSrcClass * gclass)
 +gst_jack_audio_src_init (GstJackAudioSrc * src)
  {
    //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
    src->connect = DEFAULT_PROP_CONNECT;

diff --cc ext/jack/gstjackaudiosrc.h
Simple merge

diff --cc gst/matroska/matroska-mux.c
index 5fe25f9,2e4f739..442f64e
--- 1/gst/matroska/matroska-mux.c
--- 2/gst/matroska/matroska-mux.c
+++ b/gst/matroska/matroska-mux.c
@@@ -665,9 -699,20 +665,20 @@@ gst_matroska_mux_handle_src_event (GstP
        break;
    }
  
 -  return gst_pad_event_default (pad, event);
 +  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)
@@@ -1051,18 -1082,13 +1063,12 @@@ skip_details
    } 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_SIZE (codec_buf);
 +      context->codec_priv_size = gst_buffer_get_size (codec_buf);
        context->codec_priv = g_malloc0 (context->codec_priv_size);
 -      memcpy (context->codec_priv, GST_BUFFER_DATA (codec_buf),
 -          context->codec_priv_size);
 +      gst_buffer_extract (codec_buf, 0, context->codec_priv, -1);
      }
    } else if (!strcmp (mimetype, "video/x-theora")) {
      const GValue *streamheader;
@@@ -1108,9 -1130,11 +1110,10 @@@
  
      /* global headers may be in codec data */
      if (codec_buf != NULL) {
 -      context->codec_priv_size = GST_BUFFER_SIZE (codec_buf);
+       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);
 -      memcpy (context->codec_priv, GST_BUFFER_DATA (codec_buf),
 -          context->codec_priv_size);
 +      gst_buffer_extract (codec_buf, 0, context->codec_priv, -1);
      }
    } else if (!strcmp (mimetype, "video/x-msmpeg")) {
    msmpeg43:
@@@ -1149,11 -1173,12 +1152,12 @@@
  
        GstBuffer *codec_data_buf = g_value_peek_pointer (mdpr_data);
  
 -      priv_data_size = GST_BUFFER_SIZE (codec_data_buf);
 +      priv_data_size = gst_buffer_get_size (codec_data_buf);
        priv_data = g_malloc0 (priv_data_size);
  
 -      memcpy (priv_data, GST_BUFFER_DATA (codec_data_buf), priv_data_size);
 +      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;
      }
@@@ -1233,10 -1258,12 +1237,11 @@@ xiphN_streamheader_to_codecdata (const 
    }
  
    for (i = 0; i < bufarr->len; ++i) {
 -    memcpy (priv_data + offset, GST_BUFFER_DATA (buf[i]),
 -        GST_BUFFER_SIZE (buf[i]));
 -    offset += GST_BUFFER_SIZE (buf[i]);
 +    gst_buffer_extract (buf[i], 0, priv_data + offset, -1);
 +    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;
  
@@@ -1421,12 -1443,13 +1426,13 @@@ flac_streamheader_to_codecdata (const G
      return FALSE;
    }
  
 -  context->codec_priv = g_malloc (GST_BUFFER_SIZE (buffer) - 9);
 -  context->codec_priv_size = GST_BUFFER_SIZE (buffer) - 9;
 -  memcpy (context->codec_priv, GST_BUFFER_DATA (buffer) + 9,
 -      GST_BUFFER_SIZE (buffer) - 9);
+   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);
  
    for (i = 1; i < bufarr->len; i++) {
 +    guint old_size;
      bufval = &g_array_index (bufarr, GValue, i);
  
      if (G_VALUE_TYPE (bufval) != GST_TYPE_BUFFER) {
@@@ -1487,9 -1507,11 +1491,10 @@@ speex_streamheader_to_codecdata (const 
      return FALSE;
    }
  
 -  context->codec_priv = g_malloc (GST_BUFFER_SIZE (buffer));
 -  context->codec_priv_size = GST_BUFFER_SIZE (buffer);
 -  memcpy (context->codec_priv, GST_BUFFER_DATA (buffer),
 -      GST_BUFFER_SIZE (buffer));
+   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);
  
    bufval = &g_array_index (bufarr, GValue, 1);
  
@@@ -1818,11 -1837,13 +1810,13 @@@ gst_matroska_mux_audio_pad_setcaps (Gst
  
        GstBuffer *codec_data_buf = g_value_peek_pointer (mdpr_data);
  
 -      priv_data_size = GST_BUFFER_SIZE (codec_data_buf);
 +      priv_data_size = gst_buffer_get_size (codec_data_buf);
        priv_data = g_malloc0 (priv_data_size);
  
 -      memcpy (priv_data, GST_BUFFER_DATA (codec_data_buf), priv_data_size);
 +      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;
      }
@@@ -2030,14 -2046,12 +2021,13 @@@ gst_matroska_mux_subtitle_pad_setcaps (
        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, GST_BUFFER_DATA (buf), priv_data_size);
 +    memcpy (priv_data, priv_buffer_data, priv_data_size);
      context->codec_priv = priv_data;
      context->codec_priv_size = priv_data_size;
 +    gst_buffer_unmap (buf, priv_buffer_data, priv_data_size);
    }
  
    GST_DEBUG_OBJECT (pad, "codec_id %s, codec data size %u",

diff --cc gst/matroska/matroska-read-common.c
index 0b45855,d2c2bea..2f147ea
--- 1/gst/matroska/matroska-read-common.c
--- 2/gst/matroska/matroska-read-common.c
+++ b/gst/matroska/matroska-read-common.c
@@@ -649,7 -632,10 +649,10 @@@ gst_matroska_read_common_parse_attached
            NULL);
      else
        gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_ATTACHMENT,
 -          tagbuffer, NULL);
 +          tagsample, NULL);
+ 
 -    /* the tag list adds it own ref */
 -    gst_buffer_unref (tagbuffer);
++    /* the list adds it own ref */
++    gst_buffer_unref (tagsample);
    }
  
    g_free (filename);