X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp%2Fgstrtspsrc.c;fp=gst%2Frtsp%2Fgstrtspsrc.c;h=390fb6f8eb38dba50f4d5b675f39163af9173040;hb=f05e14ba316d67b9cdf1440d9b2129da03625a30;hp=cc97bf75f2c18201dc9315c4e7e9ba4fdf913a63;hpb=ce0723527aa37d5f4d19ef8021c0b2eb8f83b08d;p=platform%2Fupstream%2Fgst-plugins-good.git

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index cc97bf7..390fb6f 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -281,6 +281,10 @@ gst_rtsp_backchannel_get_type (void)
 #define DEFAULT_BACKCHANNEL  GST_RTSP_BACKCHANNEL_NONE
 #define DEFAULT_TEARDOWN_TIMEOUT  (100 * GST_MSECOND)
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+#define DEFAULT_START_POSITION   0
+#endif
+
 enum
 {
   PROP_0,
@@ -289,6 +293,10 @@ enum
   PROP_DEBUG,
   PROP_RETRY,
   PROP_TIMEOUT,
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  PROP_START_POSITION,
+  PROP_RESUME_POSITION,
+#endif
   PROP_TCP_TIMEOUT,
   PROP_LATENCY,
   PROP_DROP_ON_LATENCY,
@@ -495,6 +503,32 @@ cmd_to_string (guint cmd)
 }
 #endif
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+static void
+gst_rtspsrc_post_error_message (GstRTSPSrc * src, GstRTSPSrcError error_id,
+    const gchar * error_string)
+{
+  GstMessage *message;
+  GstStructure *structure;
+  gboolean ret = TRUE;
+
+  GST_ERROR_OBJECT (src, "[%d] %s", error_id, error_string);
+
+  structure = gst_structure_new ("streaming_error",
+      "error_id", G_TYPE_UINT, error_id,
+      "error_string", G_TYPE_STRING, error_string, NULL);
+
+  message =
+      gst_message_new_custom (GST_MESSAGE_ERROR, GST_OBJECT (src), structure);
+
+  ret = gst_element_post_message (GST_ELEMENT (src), message);
+  if (!ret)
+    GST_ERROR_OBJECT (src, "fail to post error message.");
+
+  return;
+}
+#endif
+
 static gboolean
 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
 {
@@ -582,7 +616,18 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
           "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
           0, G_MAXUINT64, DEFAULT_TIMEOUT,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  g_object_class_install_property (gobject_class, PROP_START_POSITION,
+      g_param_spec_uint64 ("pending-start-position", "set start position",
+          "Set start position before PLAYING request.",
+          0, G_MAXUINT64, DEFAULT_START_POSITION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_RESUME_POSITION,
+      g_param_spec_uint64 ("resume-position", "set resume position",
+          "Set resume position before PLAYING request after pause.",
+          0, G_MAXUINT64, DEFAULT_START_POSITION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
   g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
       g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
           "Fail after timeout microseconds on TCP connections (0 = disabled)",
@@ -1300,6 +1345,14 @@ gst_rtspsrc_init (GstRTSPSrc * src)
   src->debug = DEFAULT_DEBUG;
   src->retry = DEFAULT_RETRY;
   src->udp_timeout = DEFAULT_TIMEOUT;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  src->start_position = DEFAULT_START_POSITION;
+  src->is_audio_codec_supported = FALSE;
+  src->is_video_codec_supported = FALSE;
+  src->audio_codec = NULL;
+  src->video_codec = NULL;
+  src->video_frame_size = NULL;
+#endif
   gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
   src->latency = DEFAULT_LATENCY_MS;
   src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
@@ -1337,6 +1390,10 @@ gst_rtspsrc_init (GstRTSPSrc * src)
   src->version = GST_RTSP_VERSION_INVALID;
   src->teardown_timeout = DEFAULT_TEARDOWN_TIMEOUT;
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  g_mutex_init (&(src)->pause_lock);
+  g_cond_init (&(src)->open_end);
+#endif
   /* get a list of all extensions */
   src->extensions = gst_rtsp_ext_list_get ();
 
@@ -1390,6 +1447,22 @@ gst_rtspsrc_finalize (GObject * object)
 
   rtspsrc = GST_RTSPSRC (object);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  rtspsrc->is_audio_codec_supported = FALSE;
+  rtspsrc->is_video_codec_supported = FALSE;
+  if (rtspsrc->audio_codec) {
+    g_free (rtspsrc->audio_codec);
+    rtspsrc->audio_codec = NULL;
+  }
+  if (rtspsrc->video_codec) {
+    g_free (rtspsrc->video_codec);
+    rtspsrc->video_codec = NULL;
+  }
+  if (rtspsrc->video_frame_size) {
+    g_free (rtspsrc->video_frame_size);
+    rtspsrc->video_frame_size = NULL;
+  }
+#endif
   gst_rtsp_ext_list_free (rtspsrc->extensions);
   g_free (rtspsrc->conninfo.location);
   gst_rtsp_url_free (rtspsrc->conninfo.url);
@@ -1399,6 +1472,11 @@ gst_rtspsrc_finalize (GObject * object)
   g_free (rtspsrc->multi_iface);
   g_free (rtspsrc->user_agent);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  g_mutex_clear (&(rtspsrc)->pause_lock);
+  g_cond_clear (&(rtspsrc)->open_end);
+#endif
+
   if (rtspsrc->sdp) {
     gst_sdp_message_free (rtspsrc->sdp);
     rtspsrc->sdp = NULL;
@@ -1535,6 +1613,16 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
     case PROP_TIMEOUT:
       rtspsrc->udp_timeout = g_value_get_uint64 (value);
       break;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    case PROP_START_POSITION:
+      rtspsrc->start_position = g_value_get_uint64 (value);
+      break;
+    case PROP_RESUME_POSITION:
+      rtspsrc->last_pos = g_value_get_uint64 (value);
+      GST_DEBUG_OBJECT (rtspsrc, "src->last_pos value set to %" GST_TIME_FORMAT,
+          GST_TIME_ARGS (rtspsrc->last_pos));
+      break;
+#endif
     case PROP_TCP_TIMEOUT:
       gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
       break;
@@ -1704,6 +1792,14 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
     case PROP_TIMEOUT:
       g_value_set_uint64 (value, rtspsrc->udp_timeout);
       break;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    case PROP_START_POSITION:
+      g_value_set_uint64 (value, rtspsrc->start_position);
+      break;
+    case PROP_RESUME_POSITION:
+      g_value_set_uint64 (value, rtspsrc->last_pos);
+      break;
+#endif
     case PROP_TCP_TIMEOUT:
     {
       guint64 timeout;
@@ -2074,7 +2170,9 @@ gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
     GstStructure *s;
     const gchar *enc;
     PtMapItem item;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    const gchar *encoder, *mediatype;
+#endif
     pt = atoi (gst_sdp_media_get_format (media, i));
 
     GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
@@ -2093,6 +2191,51 @@ gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
       if (strcmp (enc, "X-ASF-PF") == 0)
         stream->container = TRUE;
     }
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    if ((mediatype = gst_structure_get_string (s, "media"))) {
+      GST_DEBUG_OBJECT (src, " mediatype : %s", mediatype);
+      if (!strcmp (mediatype, "video")) {
+        if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+          GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+          if ((!strcmp (encoder, "H261")) ||
+              (!strcmp (encoder, "H263")) ||
+              (!strcmp (encoder, "H263-1998"))
+              || (!strcmp (encoder, "H263-2000")) || (!strcmp (encoder, "H264"))
+              || (!strcmp (encoder, "MP4V-ES"))) {
+            src->is_video_codec_supported = TRUE;
+            GST_DEBUG_OBJECT (src, "Supported Video Codec %s", encoder);
+          } else {
+            GST_DEBUG_OBJECT (src, "Unsupported Video Codec %s", encoder);
+          }
+        }
+
+        src->video_codec = g_strdup (encoder);
+        src->video_frame_size =
+            g_strdup (gst_structure_get_string (s, "a-framesize"));
+        GST_DEBUG_OBJECT (src, "video_codec %s , video_frame_size %s ",
+            src->video_codec, src->video_frame_size);
+      } else if (!strcmp (mediatype, "audio")) {
+        if ((encoder = gst_structure_get_string (s, "encoding-name"))) {
+          GST_DEBUG_OBJECT (src, " encoder : %s", encoder);
+          if ((!strcmp (encoder, "MP4A-LATM")) ||
+              (!strcmp (encoder, "AMR")) || (!strcmp (encoder, "AMR-WB"))
+              || (!strcmp (encoder, "AMR-NB"))
+              || (!strcmp (encoder, "mpeg4-generic"))
+              || (!strcmp (encoder, "MPEG4-GENERIC"))
+              || (!strcmp (encoder, "QCELP")) || ((strstr (encoder, "G726"))
+                  || (strstr (encoder, "PCMU")))) {
+            src->is_audio_codec_supported = TRUE;
+            GST_DEBUG_OBJECT (src, "Supported Audio Codec %s", encoder);
+          } else {
+            GST_DEBUG_OBJECT (src, "Unsupported Audio Codec %s", encoder);
+          }
+        }
+
+        src->audio_codec = g_strdup (encoder);
+        GST_DEBUG_OBJECT (src, "audio_codec %s ", src->audio_codec);
+      }
+    }
+#endif
 
     /* Merge in global caps */
     /* Intersect will merge in missing fields to the current caps */
@@ -2567,6 +2710,11 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
 {
   GList *walk;
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  GST_WARNING_OBJECT (src, "Setting [%s] element state to: %s \n",
+      GST_ELEMENT_NAME (GST_ELEMENT_CAST (src)),
+      gst_element_state_get_name (state));
+#endif
   if (src->manager)
     gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
 
@@ -5556,8 +5704,13 @@ receive_error:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_SERVER,
+        "Could not receive message.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
         ("Could not receive message. (%s)", str));
+#endif
     g_free (str);
 
     gst_rtsp_message_unset (&message);
@@ -5567,8 +5720,13 @@ handle_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+        "Could not handle server message.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
         ("Could not handle server message. (%s)", str));
+#endif
     g_free (str);
     gst_rtsp_message_unset (&message);
     return GST_FLOW_ERROR;
@@ -5688,8 +5846,13 @@ connect_error:
 
     src->conninfo.connected = FALSE;
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not connect to server.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
           ("Could not connect to server. (%s)", str));
+#endif
       g_free (str);
       ret = GST_FLOW_ERROR;
     } else {
@@ -5701,8 +5864,13 @@ receive_error:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+        "Could not receive message.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
         ("Could not receive message. (%s)", str));
+#endif
     g_free (str);
     return GST_FLOW_ERROR;
   }
@@ -5713,8 +5881,14 @@ handle_request_failed:
 
     gst_rtsp_message_unset (&message);
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src,
+          GST_RTSPSRC_ERROR_SERVICE_UNAVAILABLE,
+          "Could not handle server message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not handle server message. (%s)", str));
+#endif
       g_free (str);
       ret = GST_FLOW_ERROR;
     } else {
@@ -5786,10 +5960,15 @@ no_protocols:
   {
     src->cur_protocols = 0;
     /* no transport possible, post an error and stop */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_TRANSPORT,
+        "Could not receive any UDP packets for seconds, maybe your firewall is blocking it. No other protocols to try.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
         ("Could not receive any UDP packets for %.4f seconds, maybe your "
             "firewall is blocking it. No other protocols to try.",
             gst_guint64_to_gdouble (src->udp_timeout) / 1000000.0));
+#endif
     return GST_RTSP_ERROR;
   }
 open_failed:
@@ -5836,9 +6015,38 @@ gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
 static void
 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
 {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  GstMessage *s;
+  GST_WARNING_OBJECT (src, "Got cmd %s", cmd_to_string (cmd));
+#endif
+
   switch (cmd) {
     case CMD_OPEN:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      GST_DEBUG_OBJECT (src,
+          "rtsp_duration %" GST_TIME_FORMAT
+          ", rtsp_audio_codec %s , rtsp_video_codec %s , rtsp_video_frame_size %s",
+          GST_TIME_ARGS (src->segment.duration), src->audio_codec,
+          src->video_codec, src->video_frame_size);
+
+      /* post message */
+      s = gst_message_new_element (GST_OBJECT_CAST (src),
+          gst_structure_new ("rtspsrc_properties",
+              "rtsp_duration", G_TYPE_UINT64, src->segment.duration,
+              "rtsp_audio_codec", G_TYPE_STRING, src->audio_codec,
+              "rtsp_video_codec", G_TYPE_STRING, src->video_codec,
+              "rtsp_video_frame_size", G_TYPE_STRING, src->video_frame_size,
+              NULL));
+
+      gst_element_post_message (GST_ELEMENT_CAST (src), s);
+#endif
       GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      /* rtspsrc PAUSE state should be here for parsing sdp before PAUSE state changed. */
+      g_mutex_lock (&(src)->pause_lock);
+      g_cond_signal (&(src)->open_end);
+      g_mutex_unlock (&(src)->pause_lock);
+#endif
       break;
     case CMD_PLAY:
       GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
@@ -5897,6 +6105,14 @@ gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
   switch (cmd) {
     case CMD_OPEN:
       GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      /* Ending conditional wait for pause when open fails.*/
+      g_mutex_lock (&(src)->pause_lock);
+      g_cond_signal (&(src)->open_end);
+      g_mutex_unlock (&(src)->pause_lock);
+      GST_WARNING_OBJECT (src,
+          "ending conditional wait for pause as open is failed.");
+#endif
       break;
     case CMD_PLAY:
       GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
@@ -6228,8 +6444,13 @@ no_auth_available:
   {
     /* Output an error indicating that we couldn't connect because there were
      * no supported authentication protocols */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+        "No supported authentication protocol was found");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
         ("No supported authentication protocol was found"));
+#endif
     return FALSE;
   }
 no_user_pass:
@@ -6316,8 +6537,14 @@ receive_error:
         gchar *str = gst_rtsp_strresult (res);
 
         if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+          gst_rtspsrc_post_error_message (src,
+              GST_RTSPSRC_ERROR_SERVER_DISCONNECTED,
+              "Could not receive message.");
+#else
           GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
               ("Could not receive message. (%s)", str));
+#endif
         } else {
           GST_WARNING_OBJECT (src, "receive interrupted");
         }
@@ -6396,8 +6623,13 @@ send_error:
     gchar *str = gst_rtsp_strresult (res);
 
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not send message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not send message. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "send interrupted");
     }
@@ -6511,12 +6743,22 @@ error_response:
 
     switch (response->type_data.response.code) {
       case GST_RTSP_STS_NOT_FOUND:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+        gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+            "STS NOT FOUND");
+#else
         RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, NOT_FOUND,
             "Not found");
+#endif
         break;
       case GST_RTSP_STS_UNAUTHORIZED:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+        gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_NOT_AUTHORIZED,
+            "STS NOT AUTHORIZED");
+#else
         RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, NOT_AUTHORIZED,
             "Unauthorized");
+#endif
         break;
       case GST_RTSP_STS_MOVED_PERMANENTLY:
       case GST_RTSP_STS_MOVE_TEMPORARILY:
@@ -6560,8 +6802,13 @@ error_response:
         res = GST_RTSP_OK;
         break;
       default:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+        gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+            "Got error response from Server");
+#else
         RTSP_SRC_RESPONSE_ERROR (src, response, RESOURCE, READ,
             "Unhandled error");
+#endif
         break;
     }
     /* if we return ERROR we should unset the response ourselves */
@@ -6644,14 +6891,24 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
   /* ERRORS */
 no_describe:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+        "Server does not support DESCRIBE.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
         ("Server does not support DESCRIBE."));
+#endif
     return FALSE;
   }
 no_setup:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+        "Server does not support SETUP.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
         ("Server does not support SETUP."));
+#endif
     return FALSE;
   }
 }
@@ -7391,39 +7648,66 @@ gst_rtspsrc_setup_streams_start (GstRTSPSrc * src, gboolean async)
   /* ERRORS */
 no_protocols:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_PROTOCOL,
+        "Could not connect to server, no protocols left");
+#else
     /* no transport possible, post an error and stop */
     GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
         ("Could not connect to server, no protocols left"));
+#endif
     return GST_RTSP_ERROR;
   }
 no_streams:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONTENT_NOT_FOUND,
+        "SDP contains no streams");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
         ("SDP contains no streams"));
+#endif
     return GST_RTSP_ERROR;
   }
 create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not create request.");
+#else
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request. (%s)", str));
+#endif
     g_free (str);
     goto cleanup_error;
   }
 setup_transport_failed:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not setup transport.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
         ("Could not setup transport."));
+#endif
     res = GST_RTSP_ERROR;
     goto cleanup_error;
   }
 response_error:
   {
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
     const gchar *str = gst_rtsp_status_as_text (code);
+#endif
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_UNEXPECTED_MSG,
+        "Error from Server .");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
         ("Error (%d): %s", code, GST_STR_NULL (str)));
+#endif
     res = GST_RTSP_ERROR;
     goto cleanup_error;
   }
@@ -7432,8 +7716,13 @@ send_error:
     gchar *str = gst_rtsp_strresult (res);
 
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not send message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not send message. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "send interrupted");
     }
@@ -7444,15 +7733,27 @@ nothing_to_activate:
   {
     /* none of the available error codes is really right .. */
     if (unsupported_real) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src,
+          GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+          "No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams.");
+#else
       GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
           (_("No supported stream was found. You might need to install a "
                   "GStreamer RTSP extension plugin for Real media streams.")),
           (NULL));
+#endif
     } else {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src,
+          GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+          "No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin.");
+#else
       GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
           (_("No supported stream was found. You might need to allow "
                   "more transport protocols or may otherwise be missing "
                   "the right GStreamer RTSP extension plugin.")), (NULL));
+#endif
     }
     return GST_RTSP_ERROR;
   }
@@ -7503,10 +7804,21 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
   /* we need to start playback without clipping from the position reported by
    * the server */
   segment->start = seconds;
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
+/*
+The range-min points to the start of the segment , not the current position.
+After getting the current position from MSL during normal pause/resume or during seek , we should not
+update the segment->position again with the rtp header npt timestamp.
+*/
   segment->position = seconds;
+#endif
 
   if (therange->max.type == GST_RTSP_TIME_NOW)
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    seconds = 0;
+#else
     seconds = -1;
+#endif
   else if (therange->max.type == GST_RTSP_TIME_END)
     seconds = -1;
   else
@@ -7681,6 +7993,11 @@ gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
     src->control = g_strdup (control);
   }
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  src->is_audio_codec_supported = FALSE;
+  src->is_video_codec_supported = FALSE;
+#endif
+
   /* create streams */
   n_streams = gst_sdp_message_medias_len (sdp);
   for (i = 0; i < n_streams; i++) {
@@ -7688,7 +8005,15 @@ gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
   }
 
   src->state = GST_RTSP_STATE_INIT;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  /* Check for the support for the Media codecs */
+  if ((!src->is_audio_codec_supported) && (!src->is_video_codec_supported)) {
+    GST_ERROR_OBJECT (src, "UnSupported Media Type !!!! \n");
+    goto unsupported_file_type;
+  } else {
+    GST_DEBUG_OBJECT (src, "Supported Media Type. \n");
+  }
+#endif
   /* setup streams */
   if ((res = gst_rtspsrc_setup_streams_start (src, async)) < 0)
     goto setup_failed;
@@ -7708,6 +8033,17 @@ setup_failed:
     gst_rtspsrc_cleanup (src);
     return res;
   }
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+unsupported_file_type:
+  {
+    gst_rtspsrc_post_error_message (src,
+        GST_RTSPSRC_ERROR_UNSUPPORTED_MEDIA_TYPE,
+        "No supported stream was found");
+    res = GST_RTSP_ERROR;
+    gst_rtspsrc_cleanup (src);
+    return res;
+  }
+#endif
 }
 
 static GstRTSPResult
@@ -7859,8 +8195,13 @@ restart:
   /* ERRORS */
 no_url:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_INVALID_URL,
+        "No valid RTSP URL was provided");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
         ("No valid RTSP URL was provided"));
+#endif
     goto cleanup_error;
   }
 connect_failed:
@@ -7868,8 +8209,13 @@ connect_failed:
     gchar *str = gst_rtsp_strresult (res);
 
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Failed to connect.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
           ("Failed to connect. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "connect interrupted");
     }
@@ -7880,8 +8226,13 @@ create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not create request.");
+#else
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request. (%s)", str));
+#endif
     g_free (str);
     goto cleanup_error;
   }
@@ -7899,15 +8250,25 @@ methods_error:
   }
 wrong_content_type:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_OPTION_NOT_SUPPORTED,
+        "Server does not support SDP. ");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
         ("Server does not support SDP, got %s.", respcont));
+#endif
     res = GST_RTSP_ERROR;
     goto cleanup_error;
   }
 no_describe:
   {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_METHOD_NOT_ALLOWED,
+        "Server can not provide an SDP.");
+#else
     GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
         ("Server can not provide an SDP."));
+#endif
     res = GST_RTSP_ERROR;
     goto cleanup_error;
   }
@@ -8060,8 +8421,13 @@ create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not create request.");
+#else
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request. (%s)", str));
+#endif
     g_free (str);
     goto close;
   }
@@ -8071,8 +8437,13 @@ send_error:
 
     gst_rtsp_message_unset (&request);
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not send message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not send message. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
     }
@@ -8212,7 +8583,12 @@ static gchar *
 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
 {
   gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  if (src->start_position != 0 && segment->position == 0) {
+    segment->position = src->start_position;
+    src->start_position = 0;
+  }
+#endif
   if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
     g_strlcpy (val_str, "now", sizeof (val_str));
   } else {
@@ -8223,6 +8599,9 @@ gen_range_header (GstRTSPSrc * src, GstSegment * segment)
           ((gdouble) segment->position) / GST_SECOND);
     }
   }
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  GST_DEBUG_OBJECT (src, "Range Header Added : npt=%s-", val_str);
+#endif
   return g_strdup_printf ("npt=%s-", val_str);
 }
 
@@ -8338,13 +8717,34 @@ restart:
       goto create_request_failed;
 
     if (src->need_range && src->seekable >= 0.0) {
+#ifndef TIZEN_FEATURE_RTSP_MODIFICATION
       hval = gen_range_header (src, segment);
 
       gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
 
+#endif
       /* store the newsegment event so it can be sent from the streaming thread. */
       src->need_segment = TRUE;
     }
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    else {
+/*
+ Updating position with the MSL current position as gst_rtspsrc_get_position() does not return correct position.
+*/
+      GST_DEBUG_OBJECT (src,
+          " During normal pause-resume , segment->position=%" GST_TIME_FORMAT
+          ",src->start_position=%" GST_TIME_FORMAT,
+          GST_TIME_ARGS (segment->position),
+          GST_TIME_ARGS (src->start_position));
+      segment->position = src->last_pos;
+    }
+
+/*
+ Sending the npt range request for each play request for updating the segment position properly.
+*/
+    hval = gen_range_header (src, segment);
+    gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
+#endif
 
     if (segment->rate != 1.0) {
       gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
@@ -8504,8 +8904,13 @@ create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not create request. ");
+#else
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request. (%s)", str));
+#endif
     g_free (str);
     goto done;
   }
@@ -8515,8 +8920,13 @@ send_error:
 
     gst_rtsp_message_unset (&request);
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not send message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not send message. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "PLAY interrupted");
     }
@@ -8633,8 +9043,13 @@ create_request_failed:
   {
     gchar *str = gst_rtsp_strresult (res);
 
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+    gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_BAD_REQUEST,
+        "Could not create request.");
+#else
     GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
         ("Could not create request. (%s)", str));
+#endif
     g_free (str);
     goto done;
   }
@@ -8644,8 +9059,13 @@ send_error:
 
     gst_rtsp_message_unset (&request);
     if (res != GST_RTSP_EINTR) {
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      gst_rtspsrc_post_error_message (src, GST_RTSPSRC_ERROR_CONNECTION_FAIL,
+          "Could not send message.");
+#else
       GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
           ("Could not send message. (%s)", str));
+#endif
     } else {
       GST_WARNING_OBJECT (src, "PAUSE interrupted");
     }
@@ -8877,8 +9297,14 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
 {
   GstRTSPSrc *rtspsrc;
   GstStateChangeReturn ret;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  guint64 end_time;
+#endif
 
   rtspsrc = GST_RTSPSRC (element);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  GST_WARNING_OBJECT (rtspsrc, "State change transition: %d \n", transition);
+#endif
 
   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
@@ -8919,6 +9345,18 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
       ret = GST_STATE_CHANGE_SUCCESS;
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+      /* don't change to PAUSE state before complete stream opend.
+         see gst_rtspsrc_loop_complete_cmd() */
+      g_mutex_lock (&(rtspsrc)->pause_lock);
+      end_time = g_get_monotonic_time () + 10 * G_TIME_SPAN_SECOND;
+      if (!g_cond_wait_until (&(rtspsrc)->open_end, &(rtspsrc)->pause_lock,
+              end_time)) {
+        GST_WARNING_OBJECT (rtspsrc,
+            "time out: stream opend is not completed yet..");
+      }
+      g_mutex_unlock (&(rtspsrc)->pause_lock);
+#endif
       ret = GST_STATE_CHANGE_NO_PREROLL;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING: