resolve getting content-length issue when http streaming

Change-Id: I4ab68e6f017f40ffa098e53166feb3296eded91e
Signed-off-by: Hyunil Park <hyunil46.park@samsung.com>

diff --git a/src/mm_player_priv.c b/src/mm_player_priv.c
index 11e49e7..9246d61 100755 (executable)
--- a/src/mm_player_priv.c
+++ b/src/mm_player_priv.c
@@ -2281,9 +2281,9 @@ __mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data) // @
                        }
                        break;
 
-               case GST_MESSAGE_DURATION:
+               case GST_MESSAGE_DURATION_CHANGED:
                {
-                       debug_log("GST_MESSAGE_DURATION\n");
+                       debug_log("GST_MESSAGE_DURATION_CHANGED\n");
                        ret = __mmplayer_gst_handle_duration(player, msg);
                        if (!ret)
                        {
@@ -2384,7 +2384,6 @@ __mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data) // @
 static gboolean
 __mmplayer_gst_handle_duration(mm_player_t* player, GstMessage* msg)
 {
-       GstFormat format;
        gint64 bytes = 0;
 
        MMPLAYER_FENTER();
@@ -2392,23 +2391,22 @@ __mmplayer_gst_handle_duration(mm_player_t* player, GstMessage* msg)
        return_val_if_fail(player, FALSE);
        return_val_if_fail(msg, FALSE);
 
-       gst_message_parse_duration (msg, &format, &bytes);
-
-       if (MMPLAYER_IS_HTTP_STREAMING(player) && format == GST_FORMAT_BYTES )
+       if ((MMPLAYER_IS_HTTP_STREAMING(player)) &&
+               (msg->src) && (msg->src == (GstObject *)player->pipeline->mainbin[MMPLAYER_M_SRC].gst))
        {
-               debug_log("data total size of http content: %lld", bytes);
-               player->http_content_size = bytes;
+               debug_log("msg src : [%s]", GST_ELEMENT_NAME(GST_ELEMENT_CAST(msg->src)));
+
+               if (gst_element_query_duration(GST_ELEMENT_CAST(msg->src), GST_FORMAT_BYTES, &bytes))
+               {
+                       debug_log("data total size of http content: %lld", bytes);
+                       player->http_content_size = bytes;
+               }
        }
-       else if (format == GST_FORMAT_TIME)
+       else
        {
                /* handling audio clip which has vbr. means duration is keep changing */
                _mmplayer_update_content_attrs (player, ATTR_DURATION );
        }
-       else
-       {
-               debug_warning("duration is neither BYTES or TIME");
-               return FALSE;
-       }
 
        MMPLAYER_FLEAVE();
 
@@ -6853,7 +6851,7 @@ __mmplayer_check_useful_message(mm_player_t *player, GstMessage * message)
                case GST_MESSAGE_CLOCK_LOST:
                case GST_MESSAGE_NEW_CLOCK:
                case GST_MESSAGE_ELEMENT:
-               case GST_MESSAGE_DURATION:
+               case GST_MESSAGE_DURATION_CHANGED:
                case GST_MESSAGE_ASYNC_START:
                        retval = TRUE;
                        break;
@@ -6942,7 +6940,7 @@ __mmplayer_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data)
                        #endif
                        break;
 
-               case GST_MESSAGE_DURATION:
+               case GST_MESSAGE_DURATION_CHANGED:
                        __mmplayer_gst_handle_duration(player, message);
                        break;
                case GST_MESSAGE_ASYNC_DONE:
@@ -15747,22 +15745,6 @@ __gst_send_event_to_sink( mm_player_t* player, GstEvent* event )
                        {
                                debug_log("sending event[%s] to sink element [%s] success!\n",
                                        GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
-
-                               /* rtsp case, asyn_done is not called after seek during pause state */
-                               if (MMPLAYER_IS_RTSP_STREAMING(player))
-                               {
-                                       if (strstr(GST_EVENT_TYPE_NAME(event), "seek"))
-                                       {
-                                               if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PAUSED)
-                                               {
-                                                       debug_log("RTSP seek completed, after pause state..\n");
-                                                       player->doing_seek = FALSE;
-                                                       MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
-                                               }
-
-                                       }
-                               }
-
 #ifdef TEST_ES
                                if( MMPLAYER_IS_ES_BUFF_SRC(player))
                                {