mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-04 23:46:43 +00:00
a7732a68e8
The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
394 lines
14 KiB
C
394 lines
14 KiB
C
/* 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__
|
|
|
|
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_EXPORT
|
|
GType gst_rtsp_transport_mode_get_type (void);
|
|
|
|
#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
|
|
GST_EXPORT
|
|
GType gst_rtsp_suspend_mode_get_type (void);
|
|
|
|
#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type())
|
|
GST_EXPORT
|
|
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_EXPORT
|
|
GType gst_rtsp_media_get_type (void);
|
|
|
|
/* creating the media */
|
|
|
|
GST_EXPORT
|
|
GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
|
|
|
|
GST_EXPORT
|
|
GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
|
|
|
|
GST_EXPORT
|
|
GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
|
|
GstRTSPPermissions *permissions);
|
|
|
|
GST_EXPORT
|
|
GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
|
|
|
|
GST_EXPORT
|
|
GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
|
|
|
|
GST_EXPORT
|
|
GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
|
|
|
|
GST_EXPORT
|
|
GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
|
|
|
|
GST_EXPORT
|
|
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface);
|
|
|
|
GST_EXPORT
|
|
gchar * gst_rtsp_media_get_multicast_iface (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
|
|
|
|
GST_EXPORT
|
|
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
|
|
|
|
GST_EXPORT
|
|
GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
|
|
|
|
GST_EXPORT
|
|
guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
|
|
const gchar *address, guint16 port);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock);
|
|
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode);
|
|
|
|
GST_EXPORT
|
|
GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media);
|
|
|
|
/* prepare the media for playback */
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
|
|
|
|
GST_EXPORT
|
|
GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
|
|
GstSDPInfo * info);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
|
|
|
|
|
|
/* creating streams */
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
|
|
GstElement *payloader,
|
|
GstPad *pad);
|
|
|
|
/* dealing with the media */
|
|
|
|
GST_EXPORT
|
|
GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
|
|
|
|
GST_EXPORT
|
|
GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
|
|
gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media,
|
|
GstRTSPTimeRange *range,
|
|
GstSeekFlags flags);
|
|
|
|
GST_EXPORT
|
|
GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media);
|
|
|
|
GST_EXPORT
|
|
gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
|
|
gboolean play,
|
|
GstRTSPRangeUnit unit);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
|
|
GPtrArray *transports);
|
|
|
|
GST_EXPORT
|
|
void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
|
|
GstState state);
|
|
|
|
GST_EXPORT
|
|
gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);
|
|
|
|
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref)
|
|
#endif
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_MEDIA_H__ */
|