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,
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;
}
}
+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 *
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)
name = g_strdup_printf ("dynpay%d", i);
if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
/* a stream that will dynamically create pads to provide RTP packets */
-
GST_INFO ("found dynamic element %d, %p", i, elem);
g_mutex_lock (&priv->lock);
*
* 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)
*
* 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)
g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
}
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->status = status;
+ GST_DEBUG ("setting new status to %d", status);
+ g_cond_broadcast (&priv->cond);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaStatus result;
+ gint64 end_time;
+
+ g_mutex_lock (&priv->lock);
+ end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+ /* while we are preparing, wait */
+ while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+ GST_DEBUG ("waiting for status change");
+ if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+ GST_DEBUG ("timeout, assuming error status");
+ priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+ }
+ }
+ /* could be success or error */
+ result = priv->status;
+ GST_DEBUG ("got status %d", result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
/**
* gst_rtsp_media_seek:
* @media: a #GstRTSPMedia
GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
if (priv->blocked)
media_streams_set_blocked (media, TRUE);
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);
}
static void
-gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
-{
- GstRTSPMediaPrivate *priv = media->priv;
-
- g_mutex_lock (&priv->lock);
- priv->status = status;
- GST_DEBUG ("setting new status to %d", status);
- g_cond_broadcast (&priv->cond);
- g_mutex_unlock (&priv->lock);
-}
-
-/**
- * gst_rtsp_media_get_status:
- * @media: a #GstRTSPMedia
- *
- * Get the status of @media. When @media is busy preparing, this function waits
- * until @media is prepared or in error.
- *
- * Returns: the status of @media.
- */
-GstRTSPMediaStatus
-gst_rtsp_media_get_status (GstRTSPMedia * media)
-{
- GstRTSPMediaPrivate *priv = media->priv;
- GstRTSPMediaStatus result;
- gint64 end_time;
-
- g_mutex_lock (&priv->lock);
- end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
- /* while we are preparing, wait */
- while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
- GST_DEBUG ("waiting for status change");
- if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
- GST_DEBUG ("timeout, assuming error status");
- priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
- }
- }
- /* could be success or error */
- result = priv->status;
- GST_DEBUG ("got status %d", result);
- g_mutex_unlock (&priv->lock);
-
- return result;
-}
-
-static void
stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
{
*blocked &= gst_rtsp_stream_is_blocking (stream);
stream = gst_rtsp_media_create_stream (media, pay, pad);
gst_object_unref (pay);
- g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
-
GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto not_preparing;
+
+ g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
/* we will be adding elements below that will cause ASYNC_DONE to be
* posted in the bus. We want to ignore those messages until the
* pipeline really prerolled. */
/* join the element in the PAUSED state because this callback is
* called from the streaming thread and it is PAUSED */
- gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
- priv->rtpbin, GST_STATE_PAUSED);
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_PAUSED)) {
+ GST_WARNING ("failed to join bin element");
+ }
priv->adding = FALSE;
g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+not_preparing:
+ {
+ gst_rtsp_media_remove_stream (media, stream);
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("ignore pad because we are not preparing");
+ return;
+ }
}
static void
stream = g_ptr_array_index (priv->streams, i);
- gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
- priv->rtpbin, GST_STATE_NULL);
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_NULL)) {
+ goto join_bin_failed;
+ }
}
for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
return FALSE;
+join_bin_failed:
+ {
+ GST_WARNING ("failed to join bin element");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
preroll_failed:
{
GST_WARNING ("failed to preroll pipeline");
}
}
-/**
- * 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;
+ GstRTSPMediaClass *klass;
GstBus *bus;
+ GMainContext *context;
GSource *source;
- GstRTSPMediaClass *klass;
-
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), 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 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)
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;
- /* we're preparing now */
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
+ context = (thread != NULL) ? (thread->context) : NULL;
bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
g_source_set_callback (priv->source, (GSourceFunc) bus_message,
g_object_ref (media), (GDestroyNotify) watch_destroyed);
- priv->id = g_source_attach (priv->source, thread->context);
+ priv->id = g_source_attach (priv->source, context);
/* add stuff to the bin */
gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
/* do remainder in context */
source = g_idle_source_new ();
g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
- g_source_attach (source, thread->context);
+ 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);
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);
/* we are not going to use the giving thread, so stop it. */
- gst_rtsp_thread_stop (thread);
+ if (thread)
+ gst_rtsp_thread_stop (thread);
g_rec_mutex_unlock (&priv->state_lock);
return TRUE;
}
/* 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);
}
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:
GST_DEBUG ("shutting down");
+ /* release the lock on shutdown, otherwise pad_added_cb might try to
+ * acquire the lock and then we deadlock */
+ g_rec_mutex_unlock (&priv->state_lock);
set_state (media, GST_STATE_NULL);
+ g_rec_mutex_lock (&priv->state_lock);
remove_fakesink (priv);
for (i = 0; i < priv->streams->len; i++) {
priv->nettime = NULL;
priv->reused = TRUE;
- priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
/* when the media is not reusable, this will effectively unref the media and
* recreate it */
/* we need to go to playing again for the EOS to propagate, normally in this
* state, nothing is receiving data from us anymore so this is ok. */
set_state (media, GST_STATE_PLAYING);
- priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
} else {
finish_unprepare (media);
}
goto is_busy;
GST_INFO ("unprepare media %p", media);
+ if (priv->blocked)
+ media_streams_set_blocked (media, FALSE);
set_target_state (media, GST_STATE_NULL, FALSE);
success = TRUE;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
GstRTSPMediaClass *klass;
/**
* 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
}
}
-/**
- * gst_rtsp_media_suspend:
- * @media: a #GstRTSPMedia
- *
- * Suspend @media. The state of the pipeline managed by @media is set to
- * GST_STATE_NULL but all streams are kept. @media can be prepared again
- * with gst_rtsp_media_undo_reset()
- *
- * @media must be prepared with gst_rtsp_media_prepare();
- *
- * Returns: %TRUE on success.
- */
+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
-gst_rtsp_media_suspend (GstRTSPMedia * media)
+default_suspend (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstStateChangeReturn ret;
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
-
- GST_FIXME ("suspend for dynamic pipelines needs fixing");
-
- g_rec_mutex_lock (&priv->state_lock);
- if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
- goto not_prepared;
-
- /* don't attempt to suspend when something is busy */
- if (priv->n_active > 0)
- goto done;
-
switch (priv->suspend_mode) {
case GST_RTSP_SUSPEND_MODE_NONE:
GST_DEBUG ("media %p no suspend", 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);
- priv->status = GST_RTSP_MEDIA_STATUS_SUSPENDED;
-done:
- g_rec_mutex_unlock (&priv->state_lock);
return TRUE;
/* ERRORS */
-not_prepared:
- {
- g_rec_mutex_unlock (&priv->state_lock);
- GST_WARNING ("media %p was not prepared", media);
- return FALSE;
- }
state_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);
return FALSE;
}
}
/**
- * gst_rtsp_media_unsuspend:
+ * gst_rtsp_media_suspend:
* @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.
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
*
* Returns: %TRUE on success.
*/
gboolean
-gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+gst_rtsp_media_suspend (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
g_rec_mutex_lock (&priv->state_lock);
- if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* don't attempt to suspend when something is busy */
+ if (priv->n_active > 0)
goto done;
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->suspend) {
+ if (!klass->suspend (media))
+ goto suspend_failed;
+ }
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("media %p was not prepared", media);
+ return FALSE;
+ }
+suspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ GST_WARNING ("failed to suspend media %p", media);
+ return FALSE;
+ }
+}
+
+/* call with state_lock */
+gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
switch (priv->suspend_mode) {
case GST_RTSP_SUSPEND_MODE_NONE:
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
break;
case GST_RTSP_SUSPEND_MODE_PAUSE:
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
break;
case GST_RTSP_SUSPEND_MODE_RESET:
{
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
if (!start_preroll (media))
goto start_failed;
g_rec_mutex_unlock (&priv->state_lock);
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:
}
}
+/**
+ * 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)
projects / platform / upstream / gstreamer.git / blobdiff