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gstreamer/gst/rtsp-server/rtsp-media.c
Aleix Conchillo Flaqué 3fdae13fb7 media: add setup_sdp vmethod 
gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
gst_rtsp_media_setup_sdp.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2013-12-19 15:10:30 +01:00

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-media
* @short_description: The media pipeline
* @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
* #GstRTSPSessionMedia
*
* a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
* streaming to the clients. The actual data transfer is done by the
* #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
*
* The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
* client does a DESCRIBE or SETUP of a resource.
*
* A media is created with gst_rtsp_media_new() that takes the element that will
* provide the streaming elements. For each of the streams, a new #GstRTSPStream
* object needs to be made with the gst_rtsp_media_create_stream() which takes
* the payloader element and the source pad that produces the RTP stream.
*
* The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
* prepare method will add rtpbin and sinks and sources to send and receive RTP
* and RTCP packets from the clients. Each stream srcpad is connected to an
* input into the internal rtpbin.
*
* It is also possible to dynamically create #GstRTSPStream objects during the
* prepare phase. With gst_rtsp_media_get_status() you can check the status of
* the prepare phase.
*
* After the media is prepared, it is ready for streaming. It will usually be
* managed in a session with gst_rtsp_session_manage_media(). See
* #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().
*
* 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
* cleanly shut down.
*
* With gst_rtsp_media_set_shared(), the media can be shared between multiple
* clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
* can be prepared again after an unprepare.
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-media.h"
#define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
struct _GstRTSPMediaPrivate
{
GMutex lock;
GCond cond;
/* protected by lock */
GstRTSPPermissions *permissions;
gboolean shared;
gboolean suspend_mode;
gboolean reusable;
GstRTSPLowerTrans protocols;
gboolean reused;
gboolean eos_shutdown;
guint buffer_size;
GstRTSPAddressPool *pool;
gboolean blocked;
GstElement *element;
GRecMutex state_lock; /* locking order: state lock, lock */
GPtrArray *streams; /* protected by lock */
GList *dynamic; /* protected by lock */
GstRTSPMediaStatus status; /* protected by lock */
gint prepare_count;
gint n_active;
gboolean adding;
/* the pipeline for the media */
GstElement *pipeline;
GstElement *fakesink; /* protected by lock */
GSource *source;
guint id;
GstRTSPThread *thread;
gboolean time_provider;
GstNetTimeProvider *nettime;
gboolean is_live;
gboolean seekable;
gboolean buffering;
GstState target_state;
/* RTP session manager */
GstElement *rtpbin;
/* the range of media */
GstRTSPTimeRange range; /* protected by lock */
GstClockTime range_start;
GstClockTime range_stop;
};
#define DEFAULT_SHARED FALSE
#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
#define DEFAULT_REUSABLE FALSE
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_EOS_SHUTDOWN FALSE
#define DEFAULT_BUFFER_SIZE 0x80000
#define DEFAULT_TIME_PROVIDER FALSE
/* define to dump received RTCP packets */
#undef DUMP_STATS
enum
{
PROP_0,
PROP_SHARED,
PROP_SUSPEND_MODE,
PROP_REUSABLE,
PROP_PROTOCOLS,
PROP_EOS_SHUTDOWN,
PROP_BUFFER_SIZE,
PROP_ELEMENT,
PROP_TIME_PROVIDER,
PROP_LAST
};
enum
{
SIGNAL_NEW_STREAM,
SIGNAL_REMOVED_STREAM,
SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
SIGNAL_NEW_STATE,
SIGNAL_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
#define GST_CAT_DEFAULT rtsp_media_debug
static void gst_rtsp_media_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_finalize (GObject * obj);
static gboolean default_handle_message (GstRTSPMedia * media,
GstMessage * message);
static void finish_unprepare (GstRTSPMedia * media);
static gboolean default_unprepare (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);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
#define C_ENUM(v) ((gint) v)
#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
GType
gst_rtsp_suspend_mode_get_type (void)
{
static gsize id = 0;
static const GEnumValue values[] = {
{C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
{C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
"pause"},
{C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
"reset"},
{0, NULL, NULL}
};
if (g_once_init_enter (&id)) {
GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
g_once_init_leave (&id, tmp);
}
return (GType) id;
}
G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
GObjectClass *gobject_class;
g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_media_get_property;
gobject_class->set_property = gst_rtsp_media_set_property;
gobject_class->finalize = gst_rtsp_media_finalize;
g_object_class_install_property (gobject_class, PROP_SHARED,
g_param_spec_boolean ("shared", "Shared",
"If this media pipeline can be shared", DEFAULT_SHARED,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
g_param_spec_enum ("suspend-mode", "Suspend Mode",
"How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_REUSABLE,
g_param_spec_boolean ("reusable", "Reusable",
"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_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
"Send an EOS event to the pipeline before unpreparing",
DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
g_param_spec_uint ("buffer-size", "Buffer Size",
"The kernel UDP buffer size to use", 0, G_MAXUINT,
DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_ELEMENT,
g_param_spec_object ("element", "The Element",
"The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
g_param_spec_boolean ("time-provider", "Time Provider",
"Use a NetTimeProvider for clients",
DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_STREAM);
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);
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);
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);
GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
klass->handle_message = default_handle_message;
klass->unprepare = default_unprepare;
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
gst_rtsp_media_init (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
media->priv = priv;
priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
g_mutex_init (&priv->lock);
g_cond_init (&priv->cond);
g_rec_mutex_init (&priv->state_lock);
priv->shared = DEFAULT_SHARED;
priv->suspend_mode = DEFAULT_SUSPEND_MODE;
priv->reusable = DEFAULT_REUSABLE;
priv->protocols = DEFAULT_PROTOCOLS;
priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
priv->buffer_size = DEFAULT_BUFFER_SIZE;
priv->time_provider = DEFAULT_TIME_PROVIDER;
}
static void
gst_rtsp_media_finalize (GObject * obj)
{
GstRTSPMediaPrivate *priv;
GstRTSPMedia *media;
media = GST_RTSP_MEDIA (obj);
priv = media->priv;
GST_INFO ("finalize media %p", media);
if (priv->permissions)
gst_rtsp_permissions_unref (priv->permissions);
g_ptr_array_unref (priv->streams);
g_list_free_full (priv->dynamic, gst_object_unref);
if (priv->pipeline)
gst_object_unref (priv->pipeline);
if (priv->nettime)
gst_object_unref (priv->nettime);
gst_object_unref (priv->element);
if (priv->pool)
g_object_unref (priv->pool);
g_mutex_clear (&priv->lock);
g_cond_clear (&priv->cond);
g_rec_mutex_clear (&priv->state_lock);
G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
}
static void
gst_rtsp_media_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_ELEMENT:
g_value_set_object (value, media->priv->element);
break;
case PROP_SHARED:
g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
break;
case PROP_SUSPEND_MODE:
g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
break;
case PROP_REUSABLE:
g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
break;
case PROP_EOS_SHUTDOWN:
g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
break;
case PROP_BUFFER_SIZE:
g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
break;
case PROP_TIME_PROVIDER:
g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_media_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_ELEMENT:
media->priv->element = g_value_get_object (value);
gst_object_ref_sink (media->priv->element);
break;
case PROP_SHARED:
gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
break;
case PROP_SUSPEND_MODE:
gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
break;
case PROP_REUSABLE:
gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
break;
case PROP_PROTOCOLS:
gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
break;
case PROP_EOS_SHUTDOWN:
gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
break;
case PROP_BUFFER_SIZE:
gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
break;
case PROP_TIME_PROVIDER:
gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static gboolean
default_query_position (GstRTSPMedia * media, gint64 * position)
{
return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
position);
}
static gboolean
default_query_stop (GstRTSPMedia * media, gint64 * stop)
{
GstQuery *query;
gboolean res;
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;
}
static GstElement *
default_create_rtpbin (GstRTSPMedia * media)
{
GstElement *rtpbin;
rtpbin = gst_element_factory_make ("rtpbin", NULL);
return rtpbin;
}
/* must be called with state lock */
static void
collect_media_stats (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
gint64 position, stop;
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
return;
priv->range.unit = GST_RTSP_RANGE_NPT;
GST_INFO ("collect media stats");
if (priv->is_live) {
priv->range.min.type = GST_RTSP_TIME_NOW;
priv->range.min.seconds = -1;
priv->range_start = -1;
priv->range.max.type = GST_RTSP_TIME_END;
priv->range.max.seconds = -1;
priv->range_stop = -1;
} else {
GstRTSPMediaClass *klass;
gboolean ret;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
/* get the position */
ret = FALSE;
if (klass->query_position)
ret = klass->query_position (media, &position);
if (!ret) {
GST_INFO ("position query failed");
position = 0;
}
/* get the current segment stop */
ret = FALSE;
if (klass->query_stop)
ret = klass->query_stop (media, &stop);
if (!ret) {
GST_INFO ("stop query failed");
stop = -1;
}
GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
if (position == -1) {
priv->range.min.type = GST_RTSP_TIME_NOW;
priv->range.min.seconds = -1;
priv->range_start = -1;
} else {
priv->range.min.type = GST_RTSP_TIME_SECONDS;
priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
priv->range_start = position;
}
if (stop == -1) {
priv->range.max.type = GST_RTSP_TIME_END;
priv->range.max.seconds = -1;
priv->range_stop = -1;
} else {
priv->range.max.type = GST_RTSP_TIME_SECONDS;
priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
priv->range_stop = stop;
}
}
}
/**
* gst_rtsp_media_new:
* @element: (transfer full): a #GstElement
*
* Create a new #GstRTSPMedia instance. @element is the bin element that
* provides the different streams. The #GstRTSPMedia object contains the
* element to produce RTP data for one or more related (audio/video/..)
* streams.
*
* Ownership is taken of @element.
*
* Returns: a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)
{
GstRTSPMedia *result;
g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
return result;
}
/**
* gst_rtsp_media_get_element:
* @media: a #GstRTSPMedia
*
* Get the element that was used when constructing @media.
*
* Returns: (transfer full): a #GstElement. Unref after usage.
*/
GstElement *
gst_rtsp_media_get_element (GstRTSPMedia * media)
{
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
return gst_object_ref (media->priv->element);
}
/**
* gst_rtsp_media_take_pipeline:
* @media: a #GstRTSPMedia
* @pipeline: (transfer full): a #GstPipeline
*
* Set @pipeline as the #GstPipeline for @media. Ownership is
* taken of @pipeline.
*/
void
gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
{
GstRTSPMediaPrivate *priv;
GstElement *old;
GstNetTimeProvider *nettime;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_return_if_fail (GST_IS_PIPELINE (pipeline));
priv = media->priv;
g_mutex_lock (&priv->lock);
old = priv->pipeline;
priv->pipeline = GST_ELEMENT_CAST (pipeline);
nettime = priv->nettime;
priv->nettime = NULL;
g_mutex_unlock (&priv->lock);
if (old)
gst_object_unref (old);
if (nettime)
gst_object_unref (nettime);
gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
}
/**
* gst_rtsp_media_set_permissions:
* @media: a #GstRTSPMedia
* @permissions: a #GstRTSPPermissions
*
* Set @permissions on @media.
*/
void
gst_rtsp_media_set_permissions (GstRTSPMedia * media,
GstRTSPPermissions * permissions)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
if (priv->permissions)
gst_rtsp_permissions_unref (priv->permissions);
if ((priv->permissions = permissions))
gst_rtsp_permissions_ref (permissions);
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_get_permissions:
* @media: a #GstRTSPMedia
*
* Get the permissions object from @media.
*
* Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
*/
GstRTSPPermissions *
gst_rtsp_media_get_permissions (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
GstRTSPPermissions *result;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
priv = media->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->permissions))
gst_rtsp_permissions_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_media_set_suspend_mode:
* @media: a #GstRTSPMedia
* @mode: the new #GstRTSPSuspendMode
*
* Control how @ media will be suspended after the SDP has been generated and
* after a PAUSE request has been performed.
*
* Media must be unprepared when setting the suspend mode.
*/
void
gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
goto was_prepared;
priv->suspend_mode = mode;
g_rec_mutex_unlock (&priv->state_lock);
return;
/* ERRORS */
was_prepared:
{
GST_WARNING ("media %p was prepared", media);
g_rec_mutex_unlock (&priv->state_lock);
}
}
/**
* gst_rtsp_media_get_suspend_mode:
* @media: a #GstRTSPMedia
*
* Get how @media will be suspended.
*
* Returns: #GstRTSPSuspendMode.
*/
GstRTSPSuspendMode
gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
GstRTSPSuspendMode res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
res = priv->suspend_mode;
g_rec_mutex_unlock (&priv->state_lock);
return res;
}
/**
* gst_rtsp_media_set_shared:
* @media: a #GstRTSPMedia
* @shared: the new value
*
* Set or unset if the pipeline for @media can be shared will multiple clients.
* When @shared is %TRUE, client requests for this media will share the media
* pipeline.
*/
void
gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->shared = shared;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_is_shared:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media can be shared between multiple clients.
*
* Returns: %TRUE if the media can be shared between clients.
*/
gboolean
gst_rtsp_media_is_shared (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->shared;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_set_reusable:
* @media: a #GstRTSPMedia
* @reusable: the new value
*
* Set or unset if the pipeline for @media can be reused after the pipeline has
* been unprepared.
*/
void
gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->reusable = reusable;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_is_reusable:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media can be reused after an unprepare.
*
* Returns: %TRUE if the media can be reused
*/
gboolean
gst_rtsp_media_is_reusable (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->reusable;
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_protocols:
* @media: a #GstRTSPMedia
* @protocols: the new flags
*
* Configure the allowed lower transport for @media.
*/
void
gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->protocols = protocols;
g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_get_protocols:
* @media: a #GstRTSPMedia
*
* Get the allowed protocols of @media.
*
* Returns: a #GstRTSPLowerTrans
*/
GstRTSPLowerTrans
gst_rtsp_media_get_protocols (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
GstRTSPLowerTrans res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
GST_RTSP_LOWER_TRANS_UNKNOWN);
priv = media->priv;
g_mutex_lock (&priv->lock);
res = priv->protocols;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_set_eos_shutdown:
* @media: a #GstRTSPMedia
* @eos_shutdown: the new value
*
* Set or unset if an EOS event will be sent to the pipeline for @media before
* it is unprepared.
*/
void
gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->eos_shutdown = eos_shutdown;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_is_eos_shutdown:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media will send an EOS down the pipeline before
* unpreparing.
*
* Returns: %TRUE if the media will send EOS before unpreparing.
*/
gboolean
gst_rtsp_media_is_eos_shutdown (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->eos_shutdown;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_set_buffer_size:
* @media: a #GstRTSPMedia
* @size: the new value
*
* Set the kernel UDP buffer size.
*/
void
gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
GST_LOG_OBJECT (media, "set buffer size %u", size);
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->buffer_size = size;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_get_buffer_size:
* @media: a #GstRTSPMedia
*
* Get the kernel UDP buffer size.
*
* Returns: the kernel UDP buffer size.
*/
guint
gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
guint res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
priv = media->priv;
g_mutex_unlock (&priv->lock);
res = priv->buffer_size;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_use_time_provider:
* @media: a #GstRTSPMedia
* @time_provider: if a #GstNetTimeProvider should be used
*
* Set @media to provide a #GstNetTimeProvider.
*/
void
gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
{
GstRTSPMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->time_provider = time_provider;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_media_is_time_provider:
* @media: a #GstRTSPMedia
*
* Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
*
* Use gst_rtsp_media_get_time_provider() to get the network clock.
*
* Returns: %TRUE if @media can provide a #GstNetTimeProvider.
*/
gboolean
gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
priv = media->priv;
g_mutex_unlock (&priv->lock);
res = priv->time_provider;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_set_address_pool:
* @media: a #GstRTSPMedia
* @pool: a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @media.
*/
void
gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
GstRTSPAddressPool * pool)
{
GstRTSPMediaPrivate *priv;
GstRTSPAddressPool *old;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
GST_LOG_OBJECT (media, "set address pool %p", pool);
g_mutex_lock (&priv->lock);
if ((old = priv->pool) != pool)
priv->pool = pool ? g_object_ref (pool) : NULL;
else
old = NULL;
g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
pool);
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_media_get_address_pool:
* @media: a #GstRTSPMedia
*
* Get the #GstRTSPAddressPool used as the address pool of @media.
*
* Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
* usage.
*/
GstRTSPAddressPool *
gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
GstRTSPAddressPool *result;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
priv = media->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_media_collect_streams:
* @media: a #GstRTSPMedia
*
* Find all payloader elements, they should be named pay\%d in the
* element of @media, and create #GstRTSPStreams for them.
*
* Collect all dynamic elements, named dynpay\%d, and add them to
* the list of dynamic elements.
*/
void
gst_rtsp_media_collect_streams (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
GstElement *element, *elem;
GstPad *pad;
gint i;
gboolean have_elem;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
element = priv->element;
have_elem = TRUE;
for (i = 0; have_elem; i++) {
gchar *name;
have_elem = FALSE;
name = g_strdup_printf ("pay%d", i);
if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
GST_INFO ("found stream %d with payloader %p", i, elem);
/* take the pad of the payloader */
pad = gst_element_get_static_pad (elem, "src");
/* create the stream */
gst_rtsp_media_create_stream (media, elem, pad);
gst_object_unref (pad);
gst_object_unref (elem);
have_elem = TRUE;
}
g_free (name);
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);
priv->dynamic = g_list_prepend (priv->dynamic, elem);
g_mutex_unlock (&priv->lock);
have_elem = TRUE;
}
g_free (name);
}
}
/**
* gst_rtsp_media_create_stream:
* @media: a #GstRTSPMedia
* @payloader: a #GstElement
* @srcpad: a source #GstPad
*
* Create a new stream in @media that provides RTP data on @srcpad.
* @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.
*/
GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
GstPad * pad)
{
GstRTSPMediaPrivate *priv;
GstRTSPStream *stream;
GstPad *srcpad;
gchar *name;
gint idx;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_IS_PAD (pad), NULL);
g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
priv = media->priv;
g_mutex_lock (&priv->lock);
idx = priv->streams->len;
GST_DEBUG ("media %p: creating stream with index %d", media, idx);
name = g_strdup_printf ("src_%u", idx);
srcpad = gst_ghost_pad_new (name, pad);
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (priv->element, srcpad);
g_free (name);
stream = gst_rtsp_stream_new (idx, payloader, srcpad);
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
gst_rtsp_stream_set_protocols (stream, priv->protocols);
g_ptr_array_add (priv->streams, stream);
g_mutex_unlock (&priv->lock);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
NULL);
return stream;
}
static void
gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
{
GstRTSPMediaPrivate *priv;
GstPad *srcpad;
priv = media->priv;
g_mutex_lock (&priv->lock);
/* remove the ghostpad */
srcpad = gst_rtsp_stream_get_srcpad (stream);
gst_element_remove_pad (priv->element, srcpad);
gst_object_unref (srcpad);
/* now remove the stream */
g_object_ref (stream);
g_ptr_array_remove (priv->streams, stream);
g_mutex_unlock (&priv->lock);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
stream, NULL);
g_object_unref (stream);
}
/**
* gst_rtsp_media_n_streams:
* @media: a #GstRTSPMedia
*
* Get the number of streams in this media.
*
* Returns: The number of streams.
*/
guint
gst_rtsp_media_n_streams (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
guint res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
priv = media->priv;
g_mutex_lock (&priv->lock);
res = priv->streams->len;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_get_stream:
* @media: a #GstRTSPMedia
* @idx: the stream index
*
* 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.
*/
GstRTSPStream *
gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
{
GstRTSPMediaPrivate *priv;
GstRTSPStream *res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
priv = media->priv;
g_mutex_lock (&priv->lock);
if (idx < priv->streams->len)
res = g_ptr_array_index (priv->streams, idx);
else
res = NULL;
g_mutex_unlock (&priv->lock);
return res;
}
/**
* gst_rtsp_media_find_stream:
* @media: a #GstRTSPMedia
* @control: the control of the stream
*
* 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.
*/
GstRTSPStream *
gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
{
GstRTSPMediaPrivate *priv;
GstRTSPStream *res;
gint i;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (control != NULL, NULL);
priv = media->priv;
res = NULL;
g_mutex_lock (&priv->lock);
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *test;
test = g_ptr_array_index (priv->streams, i);
if (gst_rtsp_stream_has_control (test, control)) {
res = test;
break;
}
}
g_mutex_unlock (&priv->lock);
return res;
}
/* called with state-lock */
static gboolean
default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
GstRTSPRangeUnit unit)
{
return gst_rtsp_range_convert_units (range, unit);
}
/**
* gst_rtsp_media_get_range_string:
* @media: a #GstRTSPMedia
* @play: for the PLAY request
* @unit: the unit to use for the string
*
* 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.
*/
gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
GstRTSPRangeUnit unit)
{
GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
gchar *result;
GstRTSPTimeRange range;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (klass->convert_range != NULL, FALSE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
goto not_prepared;
g_mutex_lock (&priv->lock);
/* Update the range value with current position/duration */
collect_media_stats (media);
/* make copy */
range = priv->range;
if (!play && priv->n_active > 0) {
range.min.type = GST_RTSP_TIME_NOW;
range.min.seconds = -1;
}
g_mutex_unlock (&priv->lock);
g_rec_mutex_unlock (&priv->state_lock);
if (!klass->convert_range (media, &range, unit))
goto conversion_failed;
result = gst_rtsp_range_to_string (&range);
return result;
/* ERRORS */
not_prepared:
{
GST_WARNING ("media %p was not prepared", media);
g_rec_mutex_unlock (&priv->state_lock);
return NULL;
}
conversion_failed:
{
GST_WARNING ("range conversion to unit %d failed", unit);
return NULL;
}
}
static void
stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
{
gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
}
static void
media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
{
GstRTSPMediaPrivate *priv = media->priv;
GST_DEBUG ("media %p set blocked %d", media, blocked);
priv->blocked = blocked;
g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
}
/**
* gst_rtsp_media_seek:
* @media: a #GstRTSPMedia
* @range: a #GstRTSPTimeRange
*
* Seek the pipeline of @media to @range. @media must be prepared with
* gst_rtsp_media_prepare().
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
{
GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
GstSeekFlags flags;
gboolean res;
GstClockTime start, stop;
GstSeekType start_type, stop_type;
GstQuery *query;
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);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
goto not_prepared;
/* Update the seekable state of the pipeline in case it changed */
query = gst_query_new_seeking (GST_FORMAT_TIME);
if (gst_element_query (priv->pipeline, query)) {
GstFormat format;
gboolean seekable;
gint64 start, end;
gst_query_parse_seeking (query, &format, &seekable, &start, &end);
priv->seekable = seekable;
}
gst_query_unref (query);
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))
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));
if (priv->range_start == start)
start = GST_CLOCK_TIME_NONE;
else 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)
stop_type = GST_SEEK_TYPE_SET;
if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
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;
if (priv->blocked)
media_streams_set_blocked (media, TRUE);
res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
flags, start_type, start, stop_type, stop);
/* and block for the seek to complete */
GST_INFO ("done seeking %d", res);
g_rec_mutex_unlock (&priv->state_lock);
/* wait until pipeline is prerolled again, this will also collect stats */
if (!wait_preroll (media))
goto preroll_failed;
g_rec_mutex_lock (&priv->state_lock);
GST_INFO ("prerolled again");
} else {
GST_INFO ("no seek needed");
res = TRUE;
}
g_rec_mutex_unlock (&priv->state_lock);
return res;
/* ERRORS */
not_prepared:
{
g_rec_mutex_unlock (&priv->state_lock);
GST_INFO ("media %p is not prepared", media);
return FALSE;
}
not_seekable:
{
g_rec_mutex_unlock (&priv->state_lock);
GST_INFO ("pipeline is not seekable");
return FALSE;
}
not_supported:
{
g_rec_mutex_unlock (&priv->state_lock);
GST_WARNING ("conversion to npt not supported");
return FALSE;
}
preroll_failed:
{
GST_WARNING ("failed to preroll after seek");
return FALSE;
}
}
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);
}
static gboolean
media_streams_blocking (GstRTSPMedia * media)
{
gboolean blocking = TRUE;
g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
&blocking);
return blocking;
}
/* called with state-lock */
static gboolean
default_handle_message (GstRTSPMedia * media, GstMessage * message)
{
GstRTSPMediaPrivate *priv = media->priv;
GstMessageType type;
type = GST_MESSAGE_TYPE (message);
switch (type) {
case GST_MESSAGE_STATE_CHANGED:
break;
case GST_MESSAGE_BUFFERING:
{
gint percent;
gst_message_parse_buffering (message, &percent);
/* no state management needed for live pipelines */
if (priv->is_live)
break;
if (percent == 100) {
/* a 100% message means buffering is done */
priv->buffering = FALSE;
/* 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);
} else {
GST_INFO ("Buffering done");
}
} else {
/* buffering busy */
if (priv->buffering == FALSE) {
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);
} else {
GST_INFO ("Buffering ...");
}
}
priv->buffering = TRUE;
}
break;
}
case GST_MESSAGE_LATENCY:
{
gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
break;
}
case GST_MESSAGE_ERROR:
{
GError *gerror;
gchar *debug;
gst_message_parse_error (message, &gerror, &debug);
GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
break;
}
case GST_MESSAGE_WARNING:
{
GError *gerror;
gchar *debug;
gst_message_parse_warning (message, &gerror, &debug);
GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
break;
}
case GST_MESSAGE_ELEMENT:
{
const GstStructure *s;
s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
GST_DEBUG ("media received blocking message");
if (priv->blocked && media_streams_blocking (media)) {
GST_DEBUG ("media is blocking");
collect_media_stats (media);
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
}
}
break;
}
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_ASYNC_DONE:
if (priv->adding) {
/* when we are dynamically adding pads, the addition of the udpsrc will
* temporarily produce ASYNC_DONE messages. We have to ignore them and
* wait for the final ASYNC_DONE after everything prerolled */
GST_INFO ("%p: ignoring ASYNC_DONE", media);
} else {
GST_INFO ("%p: got ASYNC_DONE", media);
collect_media_stats (media);
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
}
break;
case GST_MESSAGE_EOS:
GST_INFO ("%p: got EOS", media);
if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
GST_DEBUG ("shutting down after EOS");
finish_unprepare (media);
}
break;
default:
GST_INFO ("%p: got message type %d (%s)", media, type,
gst_message_type_get_name (type));
break;
}
return TRUE;
}
static gboolean
bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstRTSPMediaClass *klass;
gboolean ret;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_rec_mutex_lock (&priv->state_lock);
if (klass->handle_message)
ret = klass->handle_message (media, message);
else
ret = FALSE;
g_rec_mutex_unlock (&priv->state_lock);
return ret;
}
static void
watch_destroyed (GstRTSPMedia * media)
{
GST_DEBUG_OBJECT (media, "source destroyed");
g_object_unref (media);
}
static GstElement *
find_payload_element (GstElement * payloader)
{
GstElement *pay = NULL;
if (GST_IS_BIN (payloader)) {
GstIterator *iter;
GValue item = { 0 };
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;
klass =
gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
if (klass == NULL)
continue;
if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
pay = gst_object_ref (element);
g_value_unset (&item);
break;
}
g_value_unset (&item);
}
gst_iterator_free (iter);
} else {
pay = g_object_ref (payloader);
}
return pay;
}
/* called from streaming threads */
static void
pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstRTSPStream *stream;
GstElement *pay;
/* find the real payload element */
pay = find_payload_element (element);
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);
/* 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. */
priv->adding = TRUE;
/* 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);
priv->adding = FALSE;
g_rec_mutex_unlock (&priv->state_lock);
}
static void
pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstRTSPStream *stream;
stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
if (stream == NULL)
return;
GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
g_rec_mutex_lock (&priv->state_lock);
gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
g_rec_mutex_unlock (&priv->state_lock);
gst_rtsp_media_remove_stream (media, stream);
}
static void
remove_fakesink (GstRTSPMediaPrivate * priv)
{
GstElement *fakesink;
g_mutex_lock (&priv->lock);
if ((fakesink = priv->fakesink))
gst_object_ref (fakesink);
priv->fakesink = NULL;
g_mutex_unlock (&priv->lock);
if (fakesink) {
gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
gst_element_set_state (fakesink, GST_STATE_NULL);
gst_object_unref (fakesink);
GST_INFO ("removed fakesink");
}
}
static void
no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GST_INFO ("no more pads");
remove_fakesink (priv);
}
typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
struct _DynPaySignalHandlers
{
gulong pad_added_handler;
gulong pad_removed_handler;
gulong no_more_pads_handler;
};
static gboolean
start_preroll (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstStateChangeReturn ret;
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;
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
GST_INFO ("SUCCESS state change for media %p", media);
priv->seekable = TRUE;
break;
case GST_STATE_CHANGE_ASYNC:
GST_INFO ("ASYNC state change for media %p", media);
priv->seekable = TRUE;
break;
case GST_STATE_CHANGE_NO_PREROLL:
/* we need to go to PLAYING */
GST_INFO ("NO_PREROLL state change: live media %p", media);
/* FIXME we disable seeking for live streams for now. We should perform a
* seeking query in preroll instead */
priv->seekable = FALSE;
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);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
break;
case GST_STATE_CHANGE_FAILURE:
goto state_failed;
}
return TRUE;
state_failed:
{
GST_WARNING ("failed to preroll pipeline");
return FALSE;
}
}
static gboolean
wait_preroll (GstRTSPMedia * media)
{
GstRTSPMediaStatus status;
GST_DEBUG ("wait to preroll pipeline");
/* wait until pipeline is prerolled */
status = gst_rtsp_media_get_status (media);
if (status == GST_RTSP_MEDIA_STATUS_ERROR)
goto preroll_failed;
return TRUE;
preroll_failed:
{
GST_WARNING ("failed to preroll pipeline");
return FALSE;
}
}
static gboolean
start_prepare (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
guint i;
GList *walk;
/* link streams we already have, other streams might appear when we have
* dynamic elements */
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *stream;
stream = g_ptr_array_index (priv->streams, i);
gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
priv->rtpbin, GST_STATE_NULL);
}
for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;
DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
GST_INFO ("adding callbacks for dynamic element %p", elem);
handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
(GCallback) pad_added_cb, media);
handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
(GCallback) pad_removed_cb, media);
handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
(GCallback) no_more_pads_cb, media);
g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
/* we add a fakesink here in order to make the state change async. We remove
* the fakesink again in the no-more-pads callback. */
priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
}
if (!start_preroll (media))
goto preroll_failed;
return FALSE;
preroll_failed:
{
GST_WARNING ("failed to preroll pipeline");
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
return FALSE;
}
}
/**
* 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)
{
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;
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)
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 */
priv->status = 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);
wait_status:
g_rec_mutex_unlock (&priv->state_lock);
/* now wait for all pads to be prerolled, FIXME, we should somehow be
* able to do this async so that we don't block the server thread. */
if (!wait_preroll (media))
goto preroll_failed;
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
GST_INFO ("object %p is prerolled", media);
return TRUE;
/* OK */
was_prepared:
{
GST_LOG ("media %p was prepared", media);
g_rec_mutex_unlock (&priv->state_lock);
return TRUE;
}
/* ERRORS */
not_unprepared:
{
GST_WARNING ("media %p was not unprepared", media);
priv->prepare_count--;
g_rec_mutex_unlock (&priv->state_lock);
return FALSE;
}
is_reused:
{
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:
{
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");
return FALSE;
}
preroll_failed:
{
GST_WARNING ("failed to preroll pipeline");
gst_rtsp_media_unprepare (media);
return FALSE;
}
}
/* must be called with state-lock */
static void
finish_unprepare (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
gint i;
GList *walk;
GST_DEBUG ("shutting down");
gst_element_set_state (priv->pipeline, GST_STATE_NULL);
remove_fakesink (priv);
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *stream;
GST_INFO ("Removing elements of stream %d from pipeline", i);
stream = g_ptr_array_index (priv->streams, i);
gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
}
/* remove the pad signal handlers */
for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;
DynPaySignalHandlers *handlers;
handlers =
g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
g_assert (handlers != NULL);
g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
g_signal_handler_disconnect (G_OBJECT (elem),
handlers->pad_removed_handler);
g_signal_handler_disconnect (G_OBJECT (elem),
handlers->no_more_pads_handler);
g_slice_free (DynPaySignalHandlers, handlers);
}
gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
priv->rtpbin = NULL;
if (priv->nettime)
gst_object_unref (priv->nettime);
priv->nettime = NULL;
priv->reused = TRUE;
priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
/* when the media is not reusable, this will effectively unref the media and
* recreate it */
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
/* the source has the last ref to the media */
if (priv->source) {
GST_DEBUG ("destroy source");
g_source_destroy (priv->source);
g_source_unref (priv->source);
}
if (priv->thread) {
GST_DEBUG ("stop thread");
gst_rtsp_thread_stop (priv->thread);
}
}
/* called with state-lock */
static gboolean
default_unprepare (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
if (priv->eos_shutdown) {
GST_DEBUG ("sending EOS for shutdown");
/* ref so that we don't disappear */
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;
} else {
finish_unprepare (media);
}
return TRUE;
}
/**
* gst_rtsp_media_unprepare:
* @media: a #GstRTSPMedia
*
* Unprepare @media. After this call, the media should be prepared again before
* it can be used again. If the media is set to be non-reusable, a new instance
* must be created.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_unprepare (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
gboolean success;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
goto was_unprepared;
priv->prepare_count--;
if (priv->prepare_count > 0)
goto is_busy;
GST_INFO ("unprepare media %p", media);
priv->target_state = GST_STATE_NULL;
success = TRUE;
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
GstRTSPMediaClass *klass;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
if (klass->unprepare)
success = klass->unprepare (media);
} else {
finish_unprepare (media);
}
g_rec_mutex_unlock (&priv->state_lock);
return success;
was_unprepared:
{
g_rec_mutex_unlock (&priv->state_lock);
GST_INFO ("media %p was already unprepared", media);
return TRUE;
}
is_busy:
{
GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
g_rec_mutex_unlock (&priv->state_lock);
return TRUE;
}
}
/* should be called with state-lock */
static GstClock *
get_clock_unlocked (GstRTSPMedia * media)
{
if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
GST_DEBUG_OBJECT (media, "media was not prepared");
return NULL;
}
return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
}
/**
* gst_rtsp_media_get_clock:
* @media: a #GstRTSPMedia
*
* Get the clock that is used by the pipeline in @media.
*
* @media must be prepared before this method returns a valid clock object.
*
* Returns: (transfer full): the #GstClock used by @media. unref after usage.
*/
GstClock *
gst_rtsp_media_get_clock (GstRTSPMedia * media)
{
GstClock *clock;
GstRTSPMediaPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
clock = get_clock_unlocked (media);
g_rec_mutex_unlock (&priv->state_lock);
return clock;
}
/**
* gst_rtsp_media_get_base_time:
* @media: a #GstRTSPMedia
*
* Get the base_time that is used by the pipeline in @media.
*
* @media must be prepared before this method returns a valid base_time.
*
* Returns: the base_time used by @media.
*/
GstClockTime
gst_rtsp_media_get_base_time (GstRTSPMedia * media)
{
GstClockTime result;
GstRTSPMediaPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
goto not_prepared;
result = gst_element_get_base_time (media->priv->pipeline);
g_rec_mutex_unlock (&priv->state_lock);
return result;
/* ERRORS */
not_prepared:
{
g_rec_mutex_unlock (&priv->state_lock);
GST_DEBUG_OBJECT (media, "media was not prepared");
return GST_CLOCK_TIME_NONE;
}
}
/**
* gst_rtsp_media_get_time_provider:
* @media: a #GstRTSPMedia
* @address: an address or %NULL
* @port: a port or 0
*
* Get the #GstNetTimeProvider for the clock used by @media. The time provider
* will listen on @address and @port for client time requests.
*
* Returns: (transfer full): the #GstNetTimeProvider of @media.
*/
GstNetTimeProvider *
gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
guint16 port)
{
GstRTSPMediaPrivate *priv;
GstNetTimeProvider *provider = NULL;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
if (priv->time_provider) {
if ((provider = priv->nettime) == NULL) {
GstClock *clock;
if (priv->time_provider && (clock = get_clock_unlocked (media))) {
provider = gst_net_time_provider_new (clock, address, port);
gst_object_unref (clock);
priv->nettime = provider;
}
}
}
g_rec_mutex_unlock (&priv->state_lock);
if (provider)
gst_object_ref (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_setup_sdp:
* @sdp: a #GstSDPMessage
* @info: info
* @media: a #GstRTSPMedia
*
* Add @media specific info to @sdp. @info is used to configure the connection
* information in the SDP.
*
* Returns: TRUE on success.
*/
gboolean
gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
GstSDPInfo * info)
{
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);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
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;
}
}
/**
* 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.
*/
gboolean
gst_rtsp_media_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);
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);
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);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
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:
* @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;
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:
priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
break;
case GST_RTSP_SUSPEND_MODE_PAUSE:
priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
break;
case GST_RTSP_SUSPEND_MODE_RESET:
{
priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
if (!start_preroll (media))
goto start_failed;
g_rec_mutex_unlock (&priv->state_lock);
if (!wait_preroll (media))
goto preroll_failed;
g_rec_mutex_lock (&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_WARNING ("failed to preroll pipeline");
return FALSE;
}
}
/* must be called with state-lock */
static void
media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
{
GstRTSPMediaPrivate *priv = media->priv;
if (state == GST_STATE_NULL) {
gst_rtsp_media_unprepare (media);
} else {
GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
priv->target_state = state;
/* when we are buffering, don't update the state yet, this will be done
* when buffering finishes */
if (priv->buffering) {
GST_INFO ("Buffering busy, delay state change");
} else {
if (state == GST_STATE_PLAYING)
/* make sure pads are not blocking anymore when going to PLAYING */
media_streams_set_blocked (media, FALSE);
gst_element_set_state (priv->pipeline, state);
/* and suspend after pause */
if (state == GST_STATE_PAUSED)
gst_rtsp_media_suspend (media);
}
}
}
/**
* gst_rtsp_media_set_pipeline_state:
* @media: a #GstRTSPMedia
* @state: the target state of the pipeline
*
* Set the state of the pipeline managed by @media to @state
*/
void
gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_rec_mutex_lock (&media->priv->state_lock);
media_set_pipeline_state_locked (media, state);
g_rec_mutex_unlock (&media->priv->state_lock);
}
/**
* 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
*
* Set the state of @media to @state and for the transports in @transports.
*
* @media must be prepared with gst_rtsp_media_prepare();
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
GPtrArray * transports)
{
GstRTSPMediaPrivate *priv;
gint i;
gboolean activate, deactivate, do_state;
gint old_active;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
priv = media->priv;
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 &&
priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
goto not_prepared;
/* NULL and READY are the same */
if (state == GST_STATE_READY)
state = GST_STATE_NULL;
activate = deactivate = FALSE;
GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
media);
switch (state) {
case GST_STATE_NULL:
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
if (priv->target_state == GST_STATE_PLAYING)
deactivate = TRUE;
break;
case GST_STATE_PLAYING:
/* we're going to PLAYING, activate */
activate = TRUE;
break;
default:
break;
}
old_active = priv->n_active;
for (i = 0; i < transports->len; i++) {
GstRTSPStreamTransport *trans;
/* we need a non-NULL entry in the array */
trans = g_ptr_array_index (transports, i);
if (trans == NULL)
continue;
if (activate) {
if (gst_rtsp_stream_transport_set_active (trans, TRUE))
priv->n_active++;
} else if (deactivate) {
if (gst_rtsp_stream_transport_set_active (trans, FALSE))
priv->n_active--;
}
}
/* 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)
do_state = TRUE;
else
do_state = FALSE;
GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
media, do_state);
if (priv->target_state != state) {
if (do_state)
media_set_pipeline_state_locked (media, state);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
NULL);
}
/* remember where we are */
if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
old_active != priv->n_active))
collect_media_stats (media);
g_rec_mutex_unlock (&priv->state_lock);
return TRUE;
/* ERRORS */
not_prepared:
{
GST_WARNING ("media %p was not prepared", media);
g_rec_mutex_unlock (&priv->state_lock);
return FALSE;
}
error_status:
{
GST_WARNING ("media %p in error status while changing to state %d",
media, state);
if (state == GST_STATE_NULL) {
for (i = 0; i < transports->len; i++) {
GstRTSPStreamTransport *trans;
/* we need a non-NULL entry in the array */
trans = g_ptr_array_index (transports, i);
if (trans == NULL)
continue;
gst_rtsp_stream_transport_set_active (trans, FALSE);
}
priv->n_active = 0;
}
g_rec_mutex_unlock (&priv->state_lock);
return FALSE;
}
}
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