X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp-server%2Frtsp-media.c;h=6b81e254e5dd1bdcb465c02b636869dde684d2cb;hb=376488d8c7d0a92c56065070f9003e699533c3e5;hp=b827c7279bf458b7332467cb3011dcc45b3104fa;hpb=de2a70bb10b1e0111b3cbfa5c45c374061223965;p=platform%2Fupstream%2Fgstreamer.git

diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index b827c72..6b81e25 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -174,7 +174,10 @@ static void gst_rtsp_media_finalize (GObject * obj);
 static gboolean default_handle_message (GstRTSPMedia * media,
     GstMessage * message);
 static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
 static gboolean default_unprepare (GstRTSPMedia * media);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
 static gboolean default_convert_range (GstRTSPMedia * media,
     GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
 static gboolean default_query_position (GstRTSPMedia * media,
@@ -285,27 +288,30 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
   gst_rtsp_media_signals[SIGNAL_PREPARED] =
       g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
-      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
       g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
-      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
       g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL,
-      NULL, g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
 
   gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
       g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
-      g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
 
   GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
 
   klass->handle_message = default_handle_message;
+  klass->prepare = default_prepare;
   klass->unprepare = default_unprepare;
+  klass->suspend = default_suspend;
+  klass->unsuspend = default_unsuspend;
   klass->convert_range = default_convert_range;
   klass->query_position = default_query_position;
   klass->query_stop = default_query_stop;
@@ -446,28 +452,74 @@ gst_rtsp_media_set_property (GObject * object, guint propid,
   }
 }
 
+typedef struct
+{
+  gint64 position;
+  gboolean ret;
+} DoQueryPositionData;
+
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_position (stream, &tmp)) {
+    data->position = MAX (data->position, tmp);
+    data->ret = TRUE;
+  }
+}
+
 static gboolean
 default_query_position (GstRTSPMedia * media, gint64 * position)
 {
-  return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
-      position);
+  GstRTSPMediaPrivate *priv;
+  DoQueryPositionData data;
+
+  priv = media->priv;
+
+  data.position = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+  *position = data.position;
+
+  return data.ret;
+}
+
+typedef struct
+{
+  gint64 stop;
+  gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+    data->stop = MAX (data->stop, tmp);
+    data->ret = TRUE;
+  }
 }
 
 static gboolean
 default_query_stop (GstRTSPMedia * media, gint64 * stop)
 {
-  GstQuery *query;
-  gboolean res;
+  GstRTSPMediaPrivate *priv;
+  DoQueryStopData data;
 
-  query = gst_query_new_segment (GST_FORMAT_TIME);
-  if ((res = gst_element_query (media->priv->pipeline, query))) {
-    GstFormat format;
-    gst_query_parse_segment (query, NULL, &format, NULL, stop);
-    if (format != GST_FORMAT_TIME)
-      *stop = -1;
-  }
-  gst_query_unref (query);
-  return res;
+  priv = media->priv;
+
+  data.stop = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+  *stop = data.stop;
+
+  return data.ret;
 }
 
 static GstElement *
@@ -485,7 +537,7 @@ static void
 collect_media_stats (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
-  gint64 position, stop;
+  gint64 position = 0, stop = -1;
 
   if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
       priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
@@ -1317,8 +1369,8 @@ gst_rtsp_media_n_streams (GstRTSPMedia * media)
  *
  * Retrieve the stream with index @idx from @media.
  *
- * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
- * that index did not exist.
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
  */
 GstRTSPStream *
 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
@@ -1347,8 +1399,9 @@ gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
  *
  * Find a stream in @media with @control as the control uri.
  *
- * Returns: (transfer none): the #GstRTSPStream with control uri @control
- * or %NULL when a stream with that control did not exist.
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
  */
 GstRTSPStream *
 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
@@ -1621,7 +1674,7 @@ gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
         flags |= GST_SEEK_FLAG_KEY_UNIT;
     }
 
-    /* FIXME, we only do forwards */
+    /* FIXME, we only do forwards playback, no trick modes yet */
     res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
         flags, start_type, start, stop_type, stop);
 
@@ -2136,49 +2189,17 @@ preroll_failed:
   }
 }
 
-/**
- * gst_rtsp_media_prepare:
- * @media: a #GstRTSPMedia
- * @thread: (transfer full): a #GstRTSPThread to run the bus handler or %NULL
- *
- * Prepare @media for streaming. This function will create the objects
- * to manage the streaming. A pipeline must have been set on @media with
- * gst_rtsp_media_take_pipeline().
- *
- * It will preroll the pipeline and collect vital information about the streams
- * such as the duration.
- *
- * Returns: %TRUE on success.
- */
-gboolean
-gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
 {
   GstRTSPMediaPrivate *priv;
-  GstBus *bus;
-  GSource *source;
   GstRTSPMediaClass *klass;
+  GstBus *bus;
   GMainContext *context;
-
-  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+  GSource *source;
 
   priv = media->priv;
 
-  g_rec_mutex_lock (&priv->state_lock);
-  priv->prepare_count++;
-
-  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
-      priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
-    goto was_prepared;
-
-  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
-    goto wait_status;
-
-  if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
-    goto not_unprepared;
-
-  if (!priv->reusable && priv->reused)
-    goto is_reused;
-
   klass = GST_RTSP_MEDIA_GET_CLASS (media);
 
   if (!klass->create_rtpbin)
@@ -2199,18 +2220,9 @@ gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
   if (priv->rtpbin == NULL)
     goto no_rtpbin;
 
-  GST_INFO ("preparing media %p", media);
-
-  /* reset some variables */
-  priv->is_live = FALSE;
-  priv->seekable = FALSE;
-  priv->buffering = FALSE;
   priv->thread = thread;
   context = (thread != NULL) ? (thread->context) : NULL;
 
-  /* we're preparing now */
-  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
-
   bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
 
   /* add the pipeline bus to our custom mainloop */
@@ -2231,6 +2243,80 @@ gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
   g_source_attach (source, context);
   g_source_unref (source);
 
+  return TRUE;
+
+  /* ERRORS */
+no_create_rtpbin:
+  {
+    GST_ERROR ("no create_rtpbin function");
+    g_critical ("no create_rtpbin vmethod function set");
+    return FALSE;
+  }
+no_rtpbin:
+  {
+    GST_WARNING ("no rtpbin element");
+    g_warning ("failed to create element 'rtpbin', check your installation");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_prepare:
+ * @media: a #GstRTSPMedia
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ *   bus handler or %NULL
+ *
+ * Prepare @media for streaming. This function will create the objects
+ * to manage the streaming. A pipeline must have been set on @media with
+ * gst_rtsp_media_take_pipeline().
+ *
+ * It will preroll the pipeline and collect vital information about the streams
+ * such as the duration.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  priv->prepare_count++;
+
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+      priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto was_prepared;
+
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+    goto is_preparing;
+
+  if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+    goto not_unprepared;
+
+  if (!priv->reusable && priv->reused)
+    goto is_reused;
+
+  GST_INFO ("preparing media %p", media);
+
+  /* reset some variables */
+  priv->is_live = FALSE;
+  priv->seekable = FALSE;
+  priv->buffering = FALSE;
+
+  /* we're preparing now */
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->prepare) {
+    if (!klass->prepare (media, thread))
+      goto prepare_failed;
+  }
+
 wait_status:
   g_rec_mutex_unlock (&priv->state_lock);
 
@@ -2246,6 +2332,13 @@ wait_status:
   return TRUE;
 
   /* OK */
+is_preparing:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    goto wait_status;
+  }
 was_prepared:
   {
     GST_LOG ("media %p was prepared", media);
@@ -2258,6 +2351,9 @@ was_prepared:
   /* ERRORS */
 not_unprepared:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     GST_WARNING ("media %p was not unprepared", media);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
@@ -2265,25 +2361,22 @@ not_unprepared:
   }
 is_reused:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
     GST_WARNING ("can not reuse media %p", media);
     return FALSE;
   }
-no_create_rtpbin:
-  {
-    priv->prepare_count--;
-    g_rec_mutex_unlock (&priv->state_lock);
-    GST_ERROR ("no create_rtpbin function");
-    g_critical ("no create_rtpbin vmethod function set");
-    return FALSE;
-  }
-no_rtpbin:
+prepare_failed:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
-    GST_WARNING ("no rtpbin element");
-    g_warning ("failed to create element 'rtpbin', check your installation");
+    GST_ERROR ("failed to prepare media");
     return FALSE;
   }
 preroll_failed:
@@ -2526,7 +2619,7 @@ not_prepared:
 /**
  * gst_rtsp_media_get_time_provider:
  * @media: a #GstRTSPMedia
- * @address: an address or %NULL
+ * @address: (allow-none): an address or %NULL
  * @port: a port or 0
  *
  * Get the #GstNetTimeProvider for the clock used by @media. The time provider
@@ -2620,6 +2713,60 @@ no_setup_sdp:
   }
 }
 
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+  guint16 seq_num;
+  seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+  gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
+/* call with state_lock */
+gboolean
+default_suspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  switch (priv->suspend_mode) {
+    case GST_RTSP_SUSPEND_MODE_NONE:
+      GST_DEBUG ("media %p no suspend", media);
+      break;
+    case GST_RTSP_SUSPEND_MODE_PAUSE:
+      GST_DEBUG ("media %p suspend to PAUSED", media);
+      ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      break;
+    case GST_RTSP_SUSPEND_MODE_RESET:
+      GST_DEBUG ("media %p suspend to NULL", media);
+      ret = set_target_state (media, GST_STATE_NULL, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      /* Because payloader needs to set the sequence number as
+       * monotonic, we need to preserve the sequence number
+       * after pause. (otherwise going from pause to play,  which
+       * is actually from NULL to PLAY will create a new sequence
+       * number. */
+      g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+      break;
+    default:
+      break;
+  }
+
+  /* let the streams do the state changes freely, if any */
+  media_streams_set_blocked (media, FALSE);
+
+  return TRUE;
+
+  /* ERRORS */
+state_failed:
+  {
+    GST_WARNING ("failed changing pipeline's state for media %p", media);
+    return FALSE;
+  }
+}
+
 /**
  * gst_rtsp_media_suspend:
  * @media: a #GstRTSPMedia
@@ -2636,7 +2783,7 @@ gboolean
 gst_rtsp_media_suspend (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
-  GstStateChangeReturn ret;
+  GstRTSPMediaClass *klass;
 
   g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
 
@@ -2650,27 +2797,12 @@ gst_rtsp_media_suspend (GstRTSPMedia * media)
   if (priv->n_active > 0)
     goto done;
 
-  switch (priv->suspend_mode) {
-    case GST_RTSP_SUSPEND_MODE_NONE:
-      GST_DEBUG ("media %p no suspend", media);
-      break;
-    case GST_RTSP_SUSPEND_MODE_PAUSE:
-      GST_DEBUG ("media %p suspend to PAUSED", media);
-      ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto state_failed;
-      break;
-    case GST_RTSP_SUSPEND_MODE_RESET:
-      GST_DEBUG ("media %p suspend to NULL", media);
-      ret = set_target_state (media, GST_STATE_NULL, TRUE);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto state_failed;
-      break;
-    default:
-      break;
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->suspend) {
+    if (!klass->suspend (media))
+      goto suspend_failed;
   }
-  /* let the streams do the state changes freely, if any */
-  media_streams_set_blocked (media, FALSE);
+
   gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
 done:
   g_rec_mutex_unlock (&priv->state_lock);
@@ -2684,35 +2816,21 @@ not_prepared:
     GST_WARNING ("media %p was not prepared", media);
     return FALSE;
   }
-state_failed:
+suspend_failed:
   {
     g_rec_mutex_unlock (&priv->state_lock);
     gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
-    GST_WARNING ("failed changing pipeline's state for media %p", media);
+    GST_WARNING ("failed to suspend media %p", media);
     return FALSE;
   }
 }
 
-/**
- * gst_rtsp_media_unsuspend:
- * @media: a #GstRTSPMedia
- *
- * Unsuspend @media if it was in a suspended state. This method does nothing
- * when the media was not in the suspended state.
- *
- * Returns: %TRUE on success.
- */
+/* call with state_lock */
 gboolean
-gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+default_unsuspend (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
 
-  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
-
-  g_rec_mutex_lock (&priv->state_lock);
-  if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
-    goto done;
-
   switch (priv->suspend_mode) {
     case GST_RTSP_SUSPEND_MODE_NONE:
       gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
@@ -2735,17 +2853,13 @@ gst_rtsp_media_unsuspend (GstRTSPMedia * media)
     default:
       break;
   }
-done:
-  g_rec_mutex_unlock (&priv->state_lock);
 
   return TRUE;
 
   /* ERRORS */
 start_failed:
   {
-    g_rec_mutex_unlock (&priv->state_lock);
     GST_WARNING ("failed to preroll pipeline");
-    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
     return FALSE;
   }
 preroll_failed:
@@ -2755,6 +2869,48 @@ preroll_failed:
   }
 }
 
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto done;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->unsuspend) {
+    if (!klass->unsuspend (media))
+      goto unsuspend_failed;
+  }
+
+done:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+unsuspend_failed:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("failed to unsuspend media %p", media);
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+}
+
 /* must be called with state-lock */
 static void
 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)