* #GstRTSPSession and #GstRTSPSessionMedia.
*
* The state of the media can be controlled with gst_rtsp_media_set_state ().
- * Seeking can be done with gst_rtsp_media_seek().
+ * Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
+ * or gst_rtsp_media_seek_trickmode() for finer control of the seek.
*
* With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
* gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
gint prepare_count;
gint n_active;
gboolean complete;
+ gboolean finishing_unprepare;
/* the pipeline for the media */
GstElement *pipeline;
gboolean do_retransmission; /* protected by lock */
guint latency; /* protected by lock */
GstClock *clock; /* protected by lock */
+ gboolean do_rate_control; /* protected by lock */
GstRTSPPublishClockMode publish_clock_mode;
/* Dynamic element handling */
guint nb_dynamic_elements;
guint no_more_pads_pending;
+ gboolean expected_async_done;
};
#define DEFAULT_SHARED FALSE
#define DEFAULT_STOP_ON_DISCONNECT TRUE
#define DEFAULT_MAX_MCAST_TTL 255
#define DEFAULT_BIND_MCAST_ADDRESS FALSE
+#define DEFAULT_DO_RATE_CONTROL TRUE
#define DEFAULT_DO_RETRANSMISSION FALSE
static gboolean wait_preroll (GstRTSPMedia * media);
-static GstElement *find_payload_element (GstElement * payloader);
+static GstElement *find_payload_element (GstElement * payloader, GstPad * pad);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
#define C_ENUM(v) ((gint) v)
+#define TRICKMODE_FLAGS (GST_SEEK_FLAG_TRICKMODE | GST_SEEK_FLAG_TRICKMODE_KEY_UNITS | GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED)
+
GType
gst_rtsp_suspend_mode_get_type (void)
{
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
+ priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
+ priv->expected_async_done = FALSE;
}
static void
return rtpbin;
}
-static gboolean
-is_receive_only (GstRTSPMedia * media)
-{
- GstRTSPMediaPrivate *priv = media->priv;
- gboolean recive_only = TRUE;
- guint i;
-
- for (i = 0; i < priv->streams->len; i++) {
- GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
- if (gst_rtsp_stream_is_sender (stream) ||
- !gst_rtsp_stream_is_receiver (stream)) {
- recive_only = FALSE;
- break;
- }
- }
-
- return recive_only;
-}
-
/* must be called with state lock */
static void
check_seekable (GstRTSPMedia * media)
GstRTSPMediaPrivate *priv = media->priv;
/* Update the seekable state of the pipeline in case it changed */
- if (is_receive_only (media)) {
+ if (gst_rtsp_media_is_receive_only (media)) {
/* TODO: Seeking for "receive-only"? */
priv->seekable = -1;
} else {
* Set the maximum time-to-live value of outgoing multicast packets.
*
* Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
* Get the the maximum time-to-live value of outgoing multicast packets.
*
* Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
*/
guint
gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
*
* Decide whether the multicast socket should be bound to a multicast address or
* INADDR_ANY.
+ *
+ * Since: 1.16
*/
void
gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
* Check if multicast sockets are configured to be bound to multicast addresses.
*
* Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
pad = gst_element_get_static_pad (elem, "src");
/* find the real payload element in case elem is a GstBin */
- pay = find_payload_element (elem);
+ pay = find_payload_element (elem, pad);
/* create the stream */
if (pay == NULL) {
g_mutex_lock (&priv->lock);
idx = priv->streams->len;
- GST_DEBUG ("media %p: creating stream with index %d", media, idx);
+ GST_DEBUG ("media %p: creating stream with index %d and payloader %"
+ GST_PTR_FORMAT, media, idx, payloader);
if (GST_PAD_IS_SRC (pad))
name = g_strdup_printf ("src_%u", idx);
}
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
- TRUE, NULL);
- g_object_set (appsink, "sync", FALSE, "async", FALSE, NULL);
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
+ FALSE, "buffer-list", TRUE, NULL);
data = g_new0 (AppSinkSrcData, 1);
data->appsink = appsink;
gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
+ gst_rtsp_stream_set_rate_control (stream, priv->do_rate_control);
g_ptr_array_add (priv->streams, stream);
}
}
+/**
+ * gst_rtsp_media_get_rates:
+ * @media: a #GstRTSPMedia
+ * @rate (allow-none): the rate of the current segment
+ * @applied_rate (allow-none): the applied_rate of the current segment
+ *
+ * Get the rate and applied_rate of the current segment.
+ *
+ * Returns: %FALSE if looking up the rate and applied rate failed. Otherwise
+ * %TRUE is returned and @rate and @applied_rate are set to the rate and
+ * applied_rate of the current segment.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
+ gdouble * applied_rate)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *stream;
+ gdouble save_rate, save_applied_rate;
+ gboolean result = TRUE;
+ gboolean first_stream = TRUE;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ if (!rate && !applied_rate) {
+ GST_WARNING_OBJECT (media, "rate and applied_rate are both NULL");
+ return FALSE;
+ }
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ g_assert (priv->streams->len > 0);
+ for (i = 0; i < priv->streams->len; i++) {
+ stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_complete (stream)) {
+ if (gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
+ if (first_stream) {
+ save_rate = *rate;
+ save_applied_rate = *applied_rate;
+ first_stream = FALSE;
+ } else {
+ if (save_rate != *rate || save_applied_rate != *applied_rate) {
+ /* diffrent rate or applied_rate, weird */
+ g_assert (FALSE);
+ result = FALSE;
+ break;
+ }
+ }
+ } else {
+ /* complete stream withot rate and applied_rate, weird */
+ g_assert (FALSE);
+ result = FALSE;
+ break;
+ }
+ }
+ }
+
+ if (!result) {
+ GST_WARNING_OBJECT (media,
+ "failed to obtain consistent rate and applied_rate");
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
static void
stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
{
static void
stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
{
- gst_rtsp_stream_unblock_linked (stream);
+ gst_rtsp_stream_set_blocked (stream, FALSE);
}
static void
-media_unblock_linked (GstRTSPMedia * media)
+media_unblock (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
- GST_DEBUG ("media %p unblocking linked streams", media);
+ GST_DEBUG ("media %p unblocking streams", media);
+ /* media is not blocked any longer, as it contains active streams,
+ * streams that are complete */
+ priv->blocked = FALSE;
g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
}
}
/**
- * gst_rtsp_media_seek_full:
+ * gst_rtsp_media_seek_trickmode:
* @media: a #GstRTSPMedia
* @range: (transfer none): a #GstRTSPTimeRange
* @flags: The minimal set of #GstSeekFlags to use
+ * @rate: the rate to use in the seek
+ * @trickmode_interval: The trickmode interval to use for KEY_UNITS trick mode
*
- * Seek the pipeline of @media to @range. @media must be prepared with
- * gst_rtsp_media_prepare().
+ * Seek the pipeline of @media to @range with the given @flags and @rate,
+ * and @trickmode_interval.
+ * @media must be prepared with gst_rtsp_media_prepare().
+ * In order to perform the seek operation, the pipeline must contain all
+ * needed transport parts (transport sinks).
*
* Returns: %TRUE on success.
+ *
+ * Since: 1.18
*/
gboolean
-gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
- GstSeekFlags flags)
+gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
+ GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate,
+ GstClockTime trickmode_interval)
{
GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
GstClockTime start, stop;
GstSeekType start_type, stop_type;
gint64 current_position;
+ gboolean force_seek;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- g_return_val_if_fail (range != NULL, FALSE);
- g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+ /* if there's a range then klass->convert_range must be set */
+ g_return_val_if_fail (range == NULL || klass->convert_range != NULL, FALSE);
+
+ GST_DEBUG ("flags=%x rate=%f", flags, rate);
priv = media->priv;
}
start_type = stop_type = GST_SEEK_TYPE_NONE;
+ start = stop = GST_CLOCK_TIME_NONE;
- if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
- goto not_supported;
- gst_rtsp_range_get_times (range, &start, &stop);
+ /* if caller provided a range convert it to NPT format
+ * if no range provided the seek is assumed to be the same position but with
+ * e.g. the rate changed */
+ if (range != NULL) {
+ if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
+ goto not_supported;
+ gst_rtsp_range_get_times (range, &start, &stop);
- GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
- GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
- GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
- GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
+ GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+ GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
+ }
current_position = -1;
if (klass->query_position)
if (start != GST_CLOCK_TIME_NONE)
start_type = GST_SEEK_TYPE_SET;
- if (priv->range_stop == stop)
- stop = GST_CLOCK_TIME_NONE;
- else if (stop != GST_CLOCK_TIME_NONE)
+ if (stop != GST_CLOCK_TIME_NONE)
stop_type = GST_SEEK_TYPE_SET;
- if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
- gboolean had_flags = flags != 0;
+ /* we force a seek if any trickmode flag is set, or if the rate
+ * is non-standard, i.e. not 1.0 */
+ force_seek = (flags & TRICKMODE_FLAGS) || rate != 1.0;
+ if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE || force_seek) {
GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
/* depends on the current playing state of the pipeline. We might need to
* queue this until we get EOS. */
- if (had_flags)
- flags |= GST_SEEK_FLAG_FLUSH;
- else
- flags = GST_SEEK_FLAG_FLUSH;
-
+ flags |= GST_SEEK_FLAG_FLUSH;
/* if range start was not supplied we must continue from current position.
* but since we're doing a flushing seek, let us query the current position
* so we end up at exactly the same position after the seek. */
- if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
+ if (range == NULL || range->min.type == GST_RTSP_TIME_END) {
if (current_position == -1) {
GST_WARNING ("current position unknown");
} else {
GST_TIME_ARGS (current_position));
start = current_position;
start_type = GST_SEEK_TYPE_SET;
- if (!had_flags)
- flags |= GST_SEEK_FLAG_ACCURATE;
}
- } else {
- /* only set keyframe flag when modifying start */
- if (start_type != GST_SEEK_TYPE_NONE)
- if (!had_flags)
- flags |= GST_SEEK_FLAG_KEY_UNIT;
}
- if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
- GST_DEBUG ("not seeking because no position change");
+ if (start == current_position && stop_type == GST_SEEK_TYPE_NONE &&
+ !force_seek) {
+ GST_DEBUG ("no position change, no flags set by caller, so not seeking");
res = TRUE;
} else {
+ GstEvent *seek_event;
+ gboolean unblock = FALSE;
+
+ /* Handle expected async-done before waiting on next async-done.
+ *
+ * Since the seek further down in code will cause a preroll and
+ * a async-done will be generated it's important to wait on async-done
+ * if that is expected. Otherwise there is the risk that the waiting
+ * for async-done after the seek is detecting the expected async-done
+ * instead of the one that corresponds to the seek. Then execution
+ * continue and act as if the pipeline is prerolled, but it's not.
+ *
+ * During wait_preroll message GST_MESSAGE_ASYNC_DONE will come
+ * and then the state will change from preparing to prepared */
+ if (priv->expected_async_done) {
+ GST_DEBUG (" expected to get async-done, waiting ");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* wait until pipeline is prerolled */
+ if (!wait_preroll (media))
+ goto preroll_failed_expected_async_done;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ GST_DEBUG (" got expected async-done");
+ }
+
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
- if (priv->blocked)
+
+ if (rate < 0.0) {
+ GstClockTime temp_time = start;
+ GstSeekType temp_type = start_type;
+
+ start = stop;
+ start_type = stop_type;
+ stop = temp_time;
+ stop_type = temp_type;
+ }
+
+ seek_event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type,
+ start, stop_type, stop);
+
+ gst_event_set_seek_trickmode_interval (seek_event, trickmode_interval);
+
+ if (!media->priv->blocked) {
+ /* Prevent a race condition with multiple streams,
+ * where one stream may have time to preroll before others
+ * have even started flushing, causing async-done to be
+ * posted too early.
+ */
media_streams_set_blocked (media, TRUE);
+ unblock = TRUE;
+ }
+
+ res = gst_element_send_event (priv->pipeline, seek_event);
- /* 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);
+ if (unblock)
+ media_streams_set_blocked (media, FALSE);
/* and block for the seek to complete */
GST_INFO ("done seeking %d", res);
GST_WARNING ("failed to preroll after seek");
return FALSE;
}
+preroll_failed_expected_async_done:
+ {
+ GST_WARNING ("failed to preroll");
+ return FALSE;
+ }
}
+/**
+ * gst_rtsp_media_seek_full:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ * @flags: The minimal set of #GstSeekFlags to use
+ *
+ * Seek the pipeline of @media to @range with the given @flags.
+ * @media must be prepared with gst_rtsp_media_prepare().
+ *
+ * Returns: %TRUE on success.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
+ GstSeekFlags flags)
+{
+ return gst_rtsp_media_seek_trickmode (media, range, flags, 1.0, 0);
+}
/**
* gst_rtsp_media_seek:
gboolean
gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
{
- return gst_rtsp_media_seek_full (media, range, 0);
+ return gst_rtsp_media_seek_trickmode (media, range, GST_SEEK_FLAG_NONE,
+ 1.0, 0);
}
-
static void
stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
{
GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
gst_element_state_get_name (old), gst_element_state_get_name (new),
gst_element_state_get_name (pending));
- if (priv->no_more_pads_pending == 0 && is_receive_only (media) &&
- old == GST_STATE_READY && new == GST_STATE_PAUSED) {
+ if (priv->no_more_pads_pending == 0
+ && gst_rtsp_media_is_receive_only (media) && old == GST_STATE_READY
+ && new == GST_STATE_PAUSED) {
GST_INFO ("%p: went to PAUSED, prepared now", media);
collect_media_stats (media);
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_ASYNC_DONE:
+ if (priv->expected_async_done)
+ priv->expected_async_done = FALSE;
if (priv->complete) {
/* receive the final ASYNC_DONE, that is posted by the media pipeline
* after all the transport parts have been successfully added to
g_object_unref (media);
}
+static gboolean
+is_payloader (GstElement * element)
+{
+ GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+ const gchar *klass;
+
+ klass = gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+ if (klass == NULL)
+ return FALSE;
+
+ if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+ return TRUE;
+ }
+
+ return FALSE;
+}
+
static GstElement *
-find_payload_element (GstElement * payloader)
+find_payload_element (GstElement * payloader, GstPad * pad)
{
GstElement *pay = NULL;
if (GST_IS_BIN (payloader)) {
GstIterator *iter;
GValue item = { 0 };
+ gchar *pad_name, *payloader_name;
+ GstElement *element;
+
+ if ((element = gst_bin_get_by_name (GST_BIN (payloader), "pay"))) {
+ if (is_payloader (element))
+ return element;
+ gst_object_unref (element);
+ }
+
+ pad_name = gst_object_get_name (GST_OBJECT (pad));
+ payloader_name = g_strdup_printf ("pay_%s", pad_name);
+ g_free (pad_name);
+ if ((element = gst_bin_get_by_name (GST_BIN (payloader), payloader_name))) {
+ g_free (payloader_name);
+ if (is_payloader (element))
+ return element;
+ gst_object_unref (element);
+ } else {
+ g_free (payloader_name);
+ }
iter = gst_bin_iterate_recurse (GST_BIN (payloader));
while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
- GstElement *element = (GstElement *) g_value_get_object (&item);
- GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
- const gchar *klass;
+ element = (GstElement *) g_value_get_object (&item);
- klass =
- gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
- if (klass == NULL)
- continue;
-
- if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+ if (is_payloader (element)) {
pay = gst_object_ref (element);
g_value_unset (&item);
break;
GstElement *pay;
/* find the real payload element */
- pay = find_payload_element (element);
+ pay = find_payload_element (element, pad);
stream = gst_rtsp_media_create_stream (media, pay, pad);
gst_object_unref (pay);
g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
}
- if (priv->nb_dynamic_elements == 0 && is_receive_only (media)) {
+ if (priv->nb_dynamic_elements == 0 && gst_rtsp_media_is_receive_only (media)) {
/* If we are receive_only (RECORD), do not try to preroll, to avoid
* a second ASYNC state change failing */
priv->is_live = TRUE;
gint i;
GList *walk;
+ if (priv->finishing_unprepare)
+ return;
+ priv->finishing_unprepare = TRUE;
+
GST_DEBUG ("shutting down");
/* release the lock on shutdown, otherwise pad_added_cb might try to
media_streams_set_blocked (media, FALSE);
- if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
- return;
-
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *stream;
GST_DEBUG ("stop thread");
gst_rtsp_thread_stop (priv->thread);
}
+
+ priv->finishing_unprepare = FALSE;
}
/* called with state-lock */
switch (priv->suspend_mode) {
case GST_RTSP_SUSPEND_MODE_NONE:
- if (is_receive_only (media))
+ if (gst_rtsp_media_is_receive_only (media))
break;
if (media_streams_blocking (media)) {
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
/* at this point the media pipeline has been updated and contain all
* specific transport parts: all active streams contain at least one sink
- * element and it's safe to unblock any blocked streams that are active */
- media_unblock_linked (media);
+ * element and it's safe to unblock all blocked streams */
+ media_unblock (media);
} else {
/* streams are not blocked and media is suspended from PAUSED */
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
/* at this point the media pipeline has been updated and contain all
* specific transport parts: all active streams contain at least one sink
- * element and it's safe to unblock any blocked streams that are active */
- media_unblock_linked (media);
+ * element and it's safe to unblock all blocked streams */
+ media_unblock (media);
if (!start_preroll (media))
goto start_failed;
media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
{
GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn set_state_ret;
+ priv->expected_async_done = FALSE;
if (state == GST_STATE_NULL) {
gst_rtsp_media_unprepare (media);
} else {
if (state == GST_STATE_PLAYING)
/* make sure pads are not blocking anymore when going to PLAYING */
- media_unblock_linked (media);
+ media_unblock (media);
- set_state (media, state);
-
- /* and suspend after pause */
- if (state == GST_STATE_PAUSED)
+ if (state == GST_STATE_PAUSED) {
+ set_state_ret = set_state (media, state);
+ if (set_state_ret == GST_STATE_CHANGE_ASYNC)
+ priv->expected_async_done = TRUE;
+ /* and suspend after pause */
gst_rtsp_media_suspend (media);
+ } else {
+ set_state (media, state);
+ }
}
}
}
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING
+ && gst_rtsp_media_is_shared (media)) {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_get_status (media);
+ g_rec_mutex_lock (&priv->state_lock);
+ }
if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
goto error_status;
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
/* we just activated the first media, do the playing state change */
if (old_active == 0 && activate)
do_state = TRUE;
- /* if we have no more active media, do the downward state changes */
- else if (priv->n_active == 0)
+ /* if we have no more active media and prepare count is not indicate
+ * that there are new session/sessions ongoing,
+ * do the downward state changes */
+ else if (priv->n_active == 0 && priv->prepare_count <= 1)
do_state = TRUE;
else
do_state = FALSE;
}
/**
- * gst_rtsp_media_get_seekable:
+ * gst_rtsp_media_seekable:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media seek and up to what point in time,
* Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
* and > 0 to indicate the longest duration between any two random access points.
* %G_MAXINT64 means any value is possible.
+ *
+ * Since: 1.14
*/
GstClockTimeDiff
gst_rtsp_media_seekable (GstRTSPMedia * media)
* SETUP.
*
* Returns: %TRUE if the media pipeline has been sucessfully updated.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
return TRUE;
}
+
+/**
+ * gst_rtsp_media_is_receive_only:
+ *
+ * Returns: %TRUE if @media is receive-only, %FALSE otherwise.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean receive_only = TRUE;
+ guint i;
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_sender (stream) ||
+ !gst_rtsp_stream_is_receiver (stream)) {
+ receive_only = FALSE;
+ break;
+ }
+ }
+
+ return receive_only;
+}
+
+/**
+ * gst_rtsp_media_has_completed_sender:
+ *
+ * See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().
+ *
+ * Returns: whether @media has at least one complete sender stream.
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gboolean sender = FALSE;
+ guint i;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_is_complete (stream))
+ if (gst_rtsp_stream_is_sender (stream) ||
+ !gst_rtsp_stream_is_receiver (stream)) {
+ sender = TRUE;
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return sender;
+}
+
+/**
+ * gst_rtsp_media_set_rate_control:
+ *
+ * Define whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+void
+gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "%s rate control", enabled ? "Enabling" : "Disabling");
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->do_rate_control = enabled;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_rate_control (stream, enabled);
+
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_rate_control:
+ *
+ * Returns: whether @media will follow the Rate-Control=no behaviour as specified
+ * in the ONVIF replay spec.
+ *
+ * Since: 1.18
+ */
+gboolean
+gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->do_rate_control;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
projects / platform / upstream / gstreamer.git / blobdiff