Add "stream-info-message" property for GSTRTSPSRC

- The rtspsrc sent stream info message when stream open is completed.
  But rtspsrc wait until stream open is completed in READY_TO_PAUSE case.
  As a result, READY_TO_PAUSE state is changed lately.
- The added "stream-info-message" property allow the user to select the
  above action.
- The default is TRUE.

[Version] 1.22.0-19
[Issue Type] Bug fix

Change-Id: I258b070156f5d485014fc2e692edffa5919383be

diff --git a/packaging/gstreamer.spec b/packaging/gstreamer.spec
index c9c83f3..1a9f4e0 100644 (file)
--- a/packaging/gstreamer.spec
+++ b/packaging/gstreamer.spec
@@ -62,7 +62,7 @@
 
 Name:           %{_name}
 Version:        1.22.0
-Release:        18
+Release:        19
 Summary:        Streaming-Media Framework Runtime
 License:        LGPL-2.0+
 Group:          Multimedia/Framework

diff --git a/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.c b/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.c
index 8e0f1d9..fecc506 100644 (file)
--- a/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.c
+++ b/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.c
@@ -315,6 +315,7 @@ gst_rtsp_backchannel_get_type (void)
 
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
 #define DEFAULT_START_POSITION   0
+#define DEFAULT_STREAM_INFO_MESSAGE   TRUE
 #endif
 
 enum
@@ -328,6 +329,7 @@ enum
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
   PROP_START_POSITION,
   PROP_RESUME_POSITION,
+  PROP_POST_STREAM_INFO_MESSAGE,
 #endif
   PROP_TCP_TIMEOUT,
   PROP_LATENCY,
@@ -665,6 +667,11 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
           "Set resume position before PLAYING request after pause.",
           0, G_MAXUINT64, DEFAULT_START_POSITION,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_POST_STREAM_INFO_MESSAGE,
+      g_param_spec_uint64 ("stream-info-message", "set stream info message",
+          "Send stream info message when stream is opened.",
+          0, G_MAXUINT64, DEFAULT_STREAM_INFO_MESSAGE,
+          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",
@@ -1796,6 +1803,10 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
       GST_DEBUG_OBJECT (rtspsrc, "src->last_pos value set to %" GST_TIME_FORMAT,
           GST_TIME_ARGS (rtspsrc->last_pos));
       break;
+    case PROP_POST_STREAM_INFO_MESSAGE:
+      rtspsrc->post_stream_info_message = g_value_get_boolean (value);
+      GST_INFO_OBJECT (rtspsrc, "src->post_stream_info_message value set to %d", rtspsrc->post_stream_info_message);
+      break;
 #endif
     case PROP_TCP_TIMEOUT:
       gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
@@ -1988,6 +1999,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
     case PROP_RESUME_POSITION:
       g_value_set_uint64 (value, rtspsrc->last_pos);
       break;
+    case PROP_POST_STREAM_INFO_MESSAGE:
+      g_value_set_boolean (value, rtspsrc->post_stream_info_message);
+      break;
 #endif
     case PROP_TCP_TIMEOUT:
       g_value_set_uint64 (value, rtspsrc->tcp_timeout);
@@ -6380,36 +6394,40 @@ static void
 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
 {
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
-  GstMessage *s;
+  GstMessage *s = NULL;
   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);
+      if (src->post_stream_info_message) {
+        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);
+      if (s) {
+        /* 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:
@@ -6470,12 +6488,14 @@ gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint 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.");
+      if (src->post_stream_info_message) {
+        /* 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:
@@ -9964,16 +9984,18 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
       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..");
+      if (rtspsrc->post_stream_info_message) {
+        /* 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);
       }
-      g_mutex_unlock (&(rtspsrc)->pause_lock);
 #endif
       if (rtspsrc->is_live)
         ret = GST_STATE_CHANGE_NO_PREROLL;

diff --git a/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.h b/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.h
index d64071e..64f98e8 100644 (file)
--- a/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.h
+++ b/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.h
@@ -392,6 +392,7 @@ struct _GstRTSPSrc {
   GMutex group_lock;
 
 #ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+  gboolean post_stream_info_message;
   GCond open_end;
   GMutex pause_lock;
   guint64 start_position;