mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 10:11:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h
Jacob Johnsson eb0272e210 rtsp-server: Add new ensure-keyunit-on-start property 
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.

With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.

Fixes #2443

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
2023-10-02 16:22:33 +00:00

466 lines
18 KiB
C
Raw Blame History

/* 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.
*/
#include <gst/gst.h>
#include <gst/rtsp/rtsp.h>
#include <gst/net/gstnet.h>
#ifndef __GST_RTSP_MEDIA_H__
#define __GST_RTSP_MEDIA_H__
#include "rtsp-server-prelude.h"
G_BEGIN_DECLS
/* types for the media */
#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
/**
* GstRTSPMediaStatus:
* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
* @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
* shutdown.
* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
* @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
*
* The state of the media pipeline.
*/
typedef enum {
GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
GST_RTSP_MEDIA_STATUS_ERROR = 5
} GstRTSPMediaStatus;
/**
* GstRTSPSuspendMode:
* @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
* @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
* @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
*
* The suspend mode of the media pipeline. A media pipeline is suspended right
* after creating the SDP and when the client performs a PAUSED request.
*/
typedef enum {
GST_RTSP_SUSPEND_MODE_NONE = 0,
GST_RTSP_SUSPEND_MODE_PAUSE = 1,
GST_RTSP_SUSPEND_MODE_RESET = 2
} GstRTSPSuspendMode;
/**
* GstRTSPTransportMode:
* @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
* @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
*
* The supported modes of the media.
*/
typedef enum {
GST_RTSP_TRANSPORT_MODE_PLAY = 1,
GST_RTSP_TRANSPORT_MODE_RECORD = 2,
} GstRTSPTransportMode;
/**
* GstRTSPPublishClockMode:
* @GST_RTSP_PUBLISH_CLOCK_MODE_NONE: Publish nothing
* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK: Publish the clock but not the offset
* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET: Publish the clock and offset
*
* Whether the clock and possibly RTP/clock offset should be published according to RFC7273.
*/
typedef enum {
GST_RTSP_PUBLISH_CLOCK_MODE_NONE,
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK,
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET
} GstRTSPPublishClockMode;
#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type())
GST_RTSP_SERVER_API
GType gst_rtsp_transport_mode_get_type (void);
#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
GST_RTSP_SERVER_API
GType gst_rtsp_suspend_mode_get_type (void);
#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type())
GST_RTSP_SERVER_API
GType gst_rtsp_publish_clock_mode_get_type (void);
#include "rtsp-stream.h"
#include "rtsp-thread-pool.h"
#include "rtsp-permissions.h"
#include "rtsp-address-pool.h"
#include "rtsp-sdp.h"
/**
* GstRTSPMedia:
*
* A class that contains the GStreamer element along with a list of
* #GstRTSPStream objects that can produce data.
*
* This object is usually created from a #GstRTSPMediaFactory.
*/
struct _GstRTSPMedia {
GObject parent;
/*< private >*/
GstRTSPMediaPrivate *priv;
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstRTSPMediaClass:
* @handle_message: handle a message
* @prepare: the default implementation adds all elements and sets the
* pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
* in case of NO_PREROLL elements).
* @unprepare: the default implementation sets the pipeline's state
* to GST_STATE_NULL and removes all elements.
* @suspend: the default implementation sets the pipeline's state to
* GST_STATE_NULL GST_STATE_PAUSED depending on the selected
* suspend mode.
* @unsuspend: the default implementation reverts the suspend operation.
* The pipeline will be prerolled again if it's state was
* set to GST_STATE_NULL in suspend.
* @convert_range: convert a range to the given unit
* @query_position: query the current position in the pipeline
* @query_stop: query when playback will stop
*
* The RTSP media class
*/
struct _GstRTSPMediaClass {
GObjectClass parent_class;
/* vmethods */
gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
gboolean (*unprepare) (GstRTSPMedia *media);
gboolean (*suspend) (GstRTSPMedia *media);
gboolean (*unsuspend) (GstRTSPMedia *media);
gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
GstRTSPRangeUnit unit);
gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
GstElement * (*create_rtpbin) (GstRTSPMedia *media);
gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
/* signals */
void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
void (*prepared) (GstRTSPMedia *media);
void (*unprepared) (GstRTSPMedia *media);
void (*target_state) (GstRTSPMedia *media, GstState state);
void (*new_state) (GstRTSPMedia *media, GstState state);
gboolean (*handle_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE-1];
};
GST_RTSP_SERVER_API
GType gst_rtsp_media_get_type (void);
/* creating the media */
GST_RTSP_SERVER_API
GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
GST_RTSP_SERVER_API
GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
GST_RTSP_SERVER_API
GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
GstRTSPPermissions *permissions);
GST_RTSP_SERVER_API
GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_can_be_shared (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
GST_RTSP_SERVER_API
GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
GST_RTSP_SERVER_API
GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
GST_RTSP_SERVER_API
GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
GST_RTSP_SERVER_API
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface);
GST_RTSP_SERVER_API
gchar * gst_rtsp_media_get_multicast_iface (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_ensure_keyunit_on_start (GstRTSPMedia* media,
gboolean ensure_keyunit_on_start);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_ensure_keyunit_on_start (GstRTSPMedia* media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_ensure_keyunit_on_start_timeout (GstRTSPMedia* media,
guint timeout);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_ensure_keyunit_on_start_timeout (GstRTSPMedia* media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
GST_RTSP_SERVER_API
GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
gboolean do_retransmission);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
GST_RTSP_SERVER_API
GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
const gchar *address, guint16 port);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode);
GST_RTSP_SERVER_API
GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia *media, gboolean bind_mcast_addr);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos);
GST_RTSP_SERVER_API
gint gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media);
/* prepare the media for playback */
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
GST_RTSP_SERVER_API
GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
GstSDPInfo * info);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
/* creating streams */
GST_RTSP_SERVER_API
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
GST_RTSP_SERVER_API
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
GstPad *pad);
/* dealing with the media */
GST_RTSP_SERVER_API
void gst_rtsp_media_lock (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_unlock (GstRTSPMedia *media);
GST_RTSP_SERVER_API
GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
GST_RTSP_SERVER_API
GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
GST_RTSP_SERVER_API
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
GST_RTSP_SERVER_API
GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
GST_RTSP_SERVER_API
GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media,
GstRTSPTimeRange *range,
GstSeekFlags flags);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_seek_trickmode (GstRTSPMedia *media,
GstRTSPTimeRange *range,
GstSeekFlags flags,
gdouble rate,
GstClockTime trickmode_interval);
GST_RTSP_SERVER_API
GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media);
GST_RTSP_SERVER_API
gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
gboolean play,
GstRTSPRangeUnit unit);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_rates (GstRTSPMedia * media,
gdouble * rate,
gdouble * applied_rate);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
GPtrArray *transports);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
GstState state);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_has_completed_sender (GstRTSPMedia * media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_rate_control (GstRTSPMedia * media);
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref)
#endif
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
Reference in a new issue View git blame Copy permalink