mirrored from git://anongit.freedesktop.org/gstreamer/gst-rtsp-server
/
rtsp-media.c
4611 lines (3816 loc) · 121 KB
/
rtsp-media.c
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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
* Copyright (C) 2015 Centricular Ltd
* Author: Sebastian Dröge <sebastian@centricular.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)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include <gst/sdp/gstmikey.h>
#include <gst/rtp/gstrtppayloads.h>
#define AES_128_KEY_LEN 16
#define AES_256_KEY_LEN 32
#define HMAC_32_KEY_LEN 4
#define HMAC_80_KEY_LEN 10
#include "rtsp-media.h"
struct _GstRTSPMediaPrivate
{
GMutex lock;
GCond cond;
/* protected by lock */
GstRTSPPermissions *permissions;
gboolean shared;
gboolean suspend_mode;
gboolean reusable;
GstRTSPProfile profiles;
GstRTSPLowerTrans protocols;
gboolean reused;
gboolean eos_shutdown;
guint buffer_size;
GstRTSPAddressPool *pool;
gchar *multicast_iface;
guint max_mcast_ttl;
gboolean bind_mcast_address;
gboolean blocked;
GstRTSPTransportMode transport_mode;
gboolean stop_on_disconnect;
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 complete;
/* the pipeline for the media */
GstElement *pipeline;
GSource *source;
guint id;
GstRTSPThread *thread;
GList *pending_pipeline_elements;
gboolean time_provider;
GstNetTimeProvider *nettime;
gboolean is_live;
GstClockTimeDiff 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;
GList *payloads; /* protected by lock */
GstClockTime rtx_time; /* protected by lock */
gboolean do_retransmission; /* protected by lock */
guint latency; /* protected by lock */
GstClock *clock; /* protected by lock */
GstRTSPPublishClockMode publish_clock_mode;
/* Dynamic element handling */
guint nb_dynamic_elements;
guint no_more_pads_pending;
};
#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
#define DEFAULT_BUFFER_SIZE 0x80000
#define DEFAULT_TIME_PROVIDER FALSE
#define DEFAULT_LATENCY 200
#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
#define DEFAULT_STOP_ON_DISCONNECT TRUE
#define DEFAULT_MAX_MCAST_TTL 255
#define DEFAULT_BIND_MCAST_ADDRESS FALSE
#define DEFAULT_DO_RETRANSMISSION FALSE
/* define to dump received RTCP packets */
#undef DUMP_STATS
enum
{
PROP_0,
PROP_SHARED,
PROP_SUSPEND_MODE,
PROP_REUSABLE,
PROP_PROFILES,
PROP_PROTOCOLS,
PROP_EOS_SHUTDOWN,
PROP_BUFFER_SIZE,
PROP_ELEMENT,
PROP_TIME_PROVIDER,
PROP_LATENCY,
PROP_TRANSPORT_MODE,
PROP_STOP_ON_DISCONNECT,
PROP_CLOCK,
PROP_MAX_MCAST_TTL,
PROP_BIND_MCAST_ADDRESS,
PROP_LAST
};
enum
{
SIGNAL_NEW_STREAM,
SIGNAL_REMOVED_STREAM,
SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
SIGNAL_TARGET_STATE,
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_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 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
static gboolean wait_preroll (GstRTSPMedia * media);
static GstElement *find_payload_element (GstElement * payloader);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
static gboolean check_complete (GstRTSPMedia * media);
#define C_ENUM(v) ((gint) v)
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;
}
#define C_FLAGS(v) ((guint) v)
GType
gst_rtsp_transport_mode_get_type (void)
{
static gsize id = 0;
static const GFlagsValue values[] = {
{C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
"play"},
{C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
"record"},
{0, NULL, NULL}
};
if (g_once_init_enter (&id)) {
GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
g_once_init_leave (&id, tmp);
}
return (GType) id;
}
GType
gst_rtsp_publish_clock_mode_get_type (void)
{
static gsize id = 0;
static const GEnumValue values[] = {
{C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
"GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
{C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
"GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
"clock"},
{C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
"GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
"clock-and-offset"},
{0, NULL, NULL}
};
if (g_once_init_enter (&id)) {
GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
g_once_init_leave (&id, tmp);
}
return (GType) id;
}
G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
GObjectClass *gobject_class;
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_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,
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));
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint ("latency", "Latency",
"Latency used for receiving media in milliseconds", 0, G_MAXUINT,
DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
g_param_spec_flags ("transport-mode", "Transport Mode",
"If this media pipeline can be used for PLAY or RECORD",
GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
"If this media pipeline should be stopped "
"when a client disconnects without TEARDOWN",
DEFAULT_STOP_ON_DISCONNECT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CLOCK,
g_param_spec_object ("clock", "Clock",
"Clock to be used by the media pipeline",
GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
"The maximum time-to-live value of outgoing multicast packets", 1,
255, DEFAULT_MAX_MCAST_TTL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
"Whether the multicast sockets should be bound to multicast addresses "
"or INADDR_ANY",
DEFAULT_BIND_MCAST_ADDRESS,
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_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_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_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;
klass->handle_sdp = default_handle_sdp;
}
static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_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->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
priv->buffer_size = DEFAULT_BUFFER_SIZE;
priv->time_provider = DEFAULT_TIME_PROVIDER;
priv->transport_mode = DEFAULT_TRANSPORT_MODE;
priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
}
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);
g_list_free_full (priv->pending_pipeline_elements, 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);
if (priv->payloads)
g_list_free (priv->payloads);
if (priv->clock)
gst_object_unref (priv->clock);
g_free (priv->multicast_iface);
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_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_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;
case PROP_LATENCY:
g_value_set_uint (value, gst_rtsp_media_get_latency (media));
break;
case PROP_TRANSPORT_MODE:
g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
break;
case PROP_STOP_ON_DISCONNECT:
g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
break;
case PROP_CLOCK:
g_value_take_object (value, gst_rtsp_media_get_clock (media));
break;
case PROP_MAX_MCAST_TTL:
g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
break;
case PROP_BIND_MCAST_ADDRESS:
g_value_set_boolean (value, gst_rtsp_media_is_bind_mcast_address (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_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;
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;
case PROP_LATENCY:
gst_rtsp_media_set_latency (media, g_value_get_uint (value));
break;
case PROP_TRANSPORT_MODE:
gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
break;
case PROP_STOP_ON_DISCONNECT:
gst_rtsp_media_set_stop_on_disconnect (media,
g_value_get_boolean (value));
break;
case PROP_CLOCK:
gst_rtsp_media_set_clock (media, g_value_get_object (value));
break;
case PROP_MAX_MCAST_TTL:
gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
break;
case PROP_BIND_MCAST_ADDRESS:
gst_rtsp_media_set_bind_mcast_address (media,
g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
typedef struct
{
gint64 position;
gboolean complete_streams_only;
gboolean ret;
} DoQueryPositionData;
static void
do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
{
gint64 tmp;
if (!gst_rtsp_stream_is_sender (stream))
return;
if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
return;
}
if (gst_rtsp_stream_query_position (stream, &tmp)) {
data->position = MIN (data->position, tmp);
data->ret = TRUE;
}
GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
GST_TIME_ARGS (data->position));
}
static gboolean
default_query_position (GstRTSPMedia * media, gint64 * position)
{
GstRTSPMediaPrivate *priv;
DoQueryPositionData data;
priv = media->priv;
data.position = G_MAXINT64;
data.ret = FALSE;
/* if the media is complete, i.e. one or more streams have been configured
* with sinks, then we want to query the position on those streams only.
* a query on an incmplete stream may return a position that originates from
* an earlier preroll */
if (check_complete (media))
data.complete_streams_only = TRUE;
else
data.complete_streams_only = FALSE;
g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
if (!data.ret)
*position = GST_CLOCK_TIME_NONE;
else
*position = data.position;
return data.ret;
}
typedef struct
{
gint64 stop;
gboolean ret;
} DoQueryStopData;
static void
do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
{
gint64 tmp = 0;
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)
{
GstRTSPMediaPrivate *priv;
DoQueryStopData data;
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 *
default_create_rtpbin (GstRTSPMedia * media)
{
GstElement *rtpbin;
rtpbin = gst_element_factory_make ("rtpbin", NULL);
return rtpbin;
}
static gboolean
is_receive_only (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
gboolean recive_only = TRUE;
guint i;
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
if (gst_rtsp_stream_is_sender (stream) ||
!gst_rtsp_stream_is_receiver (stream)) {
recive_only = FALSE;
break;
}
}
return recive_only;
}
/* must be called with state lock */
static void
check_seekable (GstRTSPMedia * media)
{
GstQuery *query;
GstRTSPMediaPrivate *priv = media->priv;
/* Update the seekable state of the pipeline in case it changed */
if (is_receive_only (media)) {
/* TODO: Seeking for "receive-only"? */
priv->seekable = -1;
} else {
guint i, n = priv->streams->len;
for (i = 0; i < n; i++) {
GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
priv->seekable = -1;
return;
}
}
}
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 ? G_MAXINT64 : 0;
} else if (priv->streams->len) {
gboolean seekable = TRUE;
guint i, n = priv->streams->len;
GST_DEBUG_OBJECT (media, "Checking %d streams", n);
for (i = 0; i < n; i++) {
GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
seekable &= gst_rtsp_stream_seekable (stream);
}
priv->seekable = seekable ? G_MAXINT64 : -1;
}
GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
gst_query_unref (query);
}
/* must be called with state lock */
static gboolean
check_complete (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
guint i, n = priv->streams->len;
for (i = 0; i < n; i++) {
GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
if (gst_rtsp_stream_is_complete (stream))
return TRUE;
}
return FALSE;
}
/* must be called with state lock */
static void
collect_media_stats (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
gint64 position = 0, stop = -1;
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;
}
check_seekable (media);
}
}
/**
* 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: (transfer full): 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;
GList *l;
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);
for (l = priv->pending_pipeline_elements; l; l = l->next) {
gst_bin_add (GST_BIN_CAST (pipeline), l->data);
}
g_list_free (priv->pending_pipeline_elements);
priv->pending_pipeline_elements = NULL;
}
/**
* gst_rtsp_media_set_permissions:
* @media: a #GstRTSPMedia
* @permissions: (transfer none) (nullable): 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);