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)
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);
/* 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);
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");
}
}
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ GstBus *bus;
+ GMainContext *context;
+ GSource *source;
+
+ priv = media->priv;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->create_rtpbin)
+ goto no_create_rtpbin;
+
+ priv->rtpbin = klass->create_rtpbin (media);
+ if (priv->rtpbin != NULL) {
+ gboolean success = TRUE;
+
+ if (klass->setup_rtpbin)
+ success = klass->setup_rtpbin (media, priv->rtpbin);
+
+ if (success == FALSE) {
+ gst_object_unref (priv->rtpbin);
+ priv->rtpbin = NULL;
+ }
+ }
+ if (priv->rtpbin == NULL)
+ goto no_rtpbin;
+
+ priv->thread = thread;
+ context = (thread != NULL) ? (thread->context) : NULL;
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+ /* add the pipeline bus to our custom mainloop */
+ priv->source = gst_bus_create_watch (bus);
+ gst_object_unref (bus);
+
+ g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+ g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+ 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, 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): a #GstRTSPThread to run the bus handler or %NULL
+ * @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_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
{
GstRTSPMediaPrivate *priv;
- GstBus *bus;
- 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;
goto was_prepared;
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
- goto wait_status;
+ goto is_preparing;
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)
- goto no_create_rtpbin;
-
- priv->rtpbin = klass->create_rtpbin (media);
- if (priv->rtpbin != NULL) {
- gboolean success = TRUE;
-
- if (klass->setup_rtpbin)
- success = klass->setup_rtpbin (media, priv->rtpbin);
-
- if (success == FALSE) {
- gst_object_unref (priv->rtpbin);
- priv->rtpbin = NULL;
- }
- }
- 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 */
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 */
- priv->source = gst_bus_create_watch (bus);
- gst_object_unref (bus);
-
- 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);
-
- /* 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_unref (source);
+ 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:
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_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
}
}
+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
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);
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);
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);
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