gboolean shared;
gboolean suspend_mode;
gboolean reusable;
+ GstRTSPProfile profiles;
GstRTSPLowerTrans protocols;
gboolean reused;
gboolean eos_shutdown;
#define DEFAULT_SHARED FALSE
#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
#define DEFAULT_REUSABLE FALSE
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_EOS_SHUTDOWN FALSE
PROP_SHARED,
PROP_SUSPEND_MODE,
PROP_REUSABLE,
+ PROP_PROFILES,
PROP_PROTOCOLS,
PROP_EOS_SHUTDOWN,
PROP_BUFFER_SIZE,
SIGNAL_REMOVED_STREAM,
SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
+ SIGNAL_TARGET_STATE,
SIGNAL_NEW_STATE,
SIGNAL_LAST
};
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,
gint64 * position);
static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
static gboolean wait_preroll (GstRTSPMedia * media);
"If this media pipeline can be reused after an unprepare",
DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
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, 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;
klass->create_rtpbin = default_create_rtpbin;
+ klass->setup_sdp = default_setup_sdp;
}
static void
priv->shared = DEFAULT_SHARED;
priv->suspend_mode = DEFAULT_SUSPEND_MODE;
priv->reusable = DEFAULT_REUSABLE;
+ priv->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
priv->buffer_size = DEFAULT_BUFFER_SIZE;
case PROP_REUSABLE:
g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+ break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
break;
case PROP_REUSABLE:
gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
break;
+ case PROP_PROFILES:
+ gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+ break;
case PROP_PROTOCOLS:
gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
break;
}
}
+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)
*
* Ownership is taken of @element.
*
- * Returns: a new #GstRTSPMedia object.
+ * Returns: (transfer full): a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)
/**
* gst_rtsp_media_set_permissions:
* @media: a #GstRTSPMedia
- * @permissions: a #GstRTSPPermissions
+ * @permissions: (transfer none): a #GstRTSPPermissions
*
* Set @permissions on @media.
*/
}
static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+ gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
{
gst_rtsp_stream_set_protocols (stream, *protocols);
/**
* gst_rtsp_media_set_address_pool:
* @media: a #GstRTSPMedia
- * @pool: a #GstRTSPAddressPool
+ * @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @media.
*/
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);
* @srcpad should be a pad of an element inside @media->element.
*
* Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
- * as @media exists.
+ * as @media exists.
*/
GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
stream = gst_rtsp_stream_new (idx, payloader, srcpad);
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
+ gst_rtsp_stream_set_profiles (stream, priv->profiles);
gst_rtsp_stream_set_protocols (stream, priv->protocols);
g_ptr_array_add (priv->streams, stream);
*
* 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)
* Get the current range as a string. @media must be prepared with
* gst_rtsp_media_prepare ().
*
- * Returns: The range as a string, g_free() after usage.
+ * Returns: (transfer full): The range as a string, g_free() after usage.
*/
gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
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
- * @range: a #GstRTSPTimeRange
+ * @range: (transfer none): a #GstRTSPTimeRange
*
* Seek the pipeline of @media to @range. @media must be prepared with
* gst_rtsp_media_prepare().
{
GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
- GstSeekFlags flags;
gboolean res;
GstClockTime start, stop;
GstSeekType start_type, stop_type;
if (!priv->seekable)
goto not_seekable;
- /* depends on the current playing state of the pipeline. We might need to
- * queue this until we get EOS. */
- flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT;
-
start_type = stop_type = GST_SEEK_TYPE_NONE;
if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
- if (priv->range_start == start)
- start = GST_CLOCK_TIME_NONE;
- else if (start != GST_CLOCK_TIME_NONE)
+ if (start != GST_CLOCK_TIME_NONE)
start_type = GST_SEEK_TYPE_SET;
if (priv->range_stop == stop)
stop_type = GST_SEEK_TYPE_SET;
if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
+ GstSeekFlags flags;
+
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);
+ /* depends on the current playing state of the pipeline. We might need to
+ * queue this until we get EOS. */
+ 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! */
+ gint64 position;
+ gboolean ret = FALSE;
+
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
+ GST_WARNING ("position query failed");
+ } else {
+ GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (position));
+ start = position;
+ start_type = GST_SEEK_TYPE_SET;
+ flags |= GST_SEEK_FLAG_ACCURATE;
+ }
+ } else {
+ /* only set keyframe flag when modifying start */
+ if (start_type != GST_SEEK_TYPE_NONE)
+ flags |= GST_SEEK_FLAG_KEY_UNIT;
+ }
+
+ /* 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);
return blocking;
}
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+ media);
+ ret = gst_element_set_state (priv->pipeline, state);
+
+ return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set target state to %s for media %p",
+ gst_element_state_get_name (state), media);
+ priv->target_state = state;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+ priv->target_state, NULL);
+
+ if (do_state)
+ ret = set_state (media, state);
+ else
+ ret = GST_STATE_CHANGE_SUCCESS;
+
+ return ret;
+}
+
/* called with state-lock */
static gboolean
default_handle_message (GstRTSPMedia * media, GstMessage * message)
/* if the desired state is playing, go back */
if (priv->target_state == GST_STATE_PLAYING) {
GST_INFO ("Buffering done, setting pipeline to PLAYING");
- gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
+ set_state (media, GST_STATE_PLAYING);
} else {
GST_INFO ("Buffering done");
}
if (priv->target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
GST_INFO ("Buffering, setting pipeline to PAUSED ...");
- gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
+ set_state (media, GST_STATE_PAUSED);
} else {
GST_INFO ("Buffering ...");
}
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
GST_INFO ("setting pipeline to PAUSED for media %p", media);
/* first go to PAUSED */
- ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
- priv->target_state = GST_STATE_PAUSED;
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
priv->is_live = TRUE;
/* start blocked to make sure nothing goes to the sink */
media_streams_set_blocked (media, TRUE);
- ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
+ ret = set_state (media, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
break;
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: 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. */
+ 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:
+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_ERROR ("no create_rtpbin function");
- g_critical ("no create_rtpbin vmethod function set");
- return FALSE;
- }
-no_rtpbin:
- {
- 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");
- gst_element_set_state (priv->pipeline, GST_STATE_NULL);
+ /* 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 */
gst_element_send_event (priv->pipeline, gst_event_new_eos ());
/* 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. */
- gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
- priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
+ set_state (media, GST_STATE_PLAYING);
} else {
finish_unprepare (media);
}
goto is_busy;
GST_INFO ("unprepare media %p", media);
- priv->target_state = GST_STATE_NULL;
+ 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
return provider;
}
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+ return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
/**
- * gst_rtsp_media_suspend:
+ * gst_rtsp_media_setup_sdp:
* @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
*
- * 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();
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
*
- * Returns: %TRUE on success.
+ * Returns: TRUE on success.
*/
gboolean
-gst_rtsp_media_suspend (GstRTSPMedia * media)
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info)
{
- GstRTSPMediaPrivate *priv = media->priv;
- GstStateChangeReturn ret;
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+ g_return_val_if_fail (info != NULL, FALSE);
- GST_FIXME ("suspend for dynamic pipelines needs fixing");
+ priv = media->priv;
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;
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->setup_sdp)
+ goto no_setup_sdp;
+
+ res = klass->setup_sdp (media, sdp, info);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_setup_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no setup_sdp function");
+ g_critical ("no setup_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+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:
break;
case GST_RTSP_SUSPEND_MODE_PAUSE:
GST_DEBUG ("media %p suspend to PAUSED", media);
- priv->target_state = GST_STATE_PAUSED;
- ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
+ 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);
- priv->target_state = GST_STATE_NULL;
- ret = gst_element_set_state (priv->pipeline, GST_STATE_NULL);
+ 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)
gst_rtsp_media_unprepare (media);
} else {
GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
- priv->target_state = state;
+ set_target_state (media, state, FALSE);
/* when we are buffering, don't update the state yet, this will be done
* when buffering finishes */
if (priv->buffering) {
/* make sure pads are not blocking anymore when going to PLAYING */
media_streams_set_blocked (media, FALSE);
- gst_element_set_state (priv->pipeline, state);
+ set_state (media, state);
/* and suspend after pause */
if (state == GST_STATE_PAUSED)
* gst_rtsp_media_set_state:
* @media: a #GstRTSPMedia
* @state: the target state of the media
- * @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
- * of #GstRTSPStreamTransport pointers
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
*
* Set the state of @media to @state and for the transports in @transports.
*
projects / platform / upstream / gstreamer.git / blobdiff