mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 05:12:09 +00:00
6459a61e8f
add dscp_qos setting support at factory and media level to setup IP DSCP field of bounded UDP sinks. Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
449 lines
17 KiB
C
449 lines
17 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtsp/rtsp.h>
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#include <gst/net/gstnet.h>
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#ifndef __GST_RTSP_MEDIA_H__
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#define __GST_RTSP_MEDIA_H__
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#include "rtsp-server-prelude.h"
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G_BEGIN_DECLS
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/* types for the media */
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#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
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#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
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#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
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#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
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#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
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#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
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#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
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#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
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typedef struct _GstRTSPMedia GstRTSPMedia;
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typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
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typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
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/**
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* GstRTSPMediaStatus:
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* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
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* @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
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* shutdown.
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* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
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* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
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* @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
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* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
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*
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* The state of the media pipeline.
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*/
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typedef enum {
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GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
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GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
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GST_RTSP_MEDIA_STATUS_PREPARING = 2,
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GST_RTSP_MEDIA_STATUS_PREPARED = 3,
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GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
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GST_RTSP_MEDIA_STATUS_ERROR = 5
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} GstRTSPMediaStatus;
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/**
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* GstRTSPSuspendMode:
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* @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
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* @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
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* @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
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*
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* The suspend mode of the media pipeline. A media pipeline is suspended right
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* after creating the SDP and when the client performs a PAUSED request.
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*/
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typedef enum {
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GST_RTSP_SUSPEND_MODE_NONE = 0,
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GST_RTSP_SUSPEND_MODE_PAUSE = 1,
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GST_RTSP_SUSPEND_MODE_RESET = 2
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} GstRTSPSuspendMode;
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/**
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* GstRTSPTransportMode:
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* @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
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* @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
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*
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* The supported modes of the media.
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*/
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typedef enum {
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GST_RTSP_TRANSPORT_MODE_PLAY = 1,
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GST_RTSP_TRANSPORT_MODE_RECORD = 2,
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} GstRTSPTransportMode;
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/**
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* GstRTSPPublishClockMode:
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* @GST_RTSP_PUBLISH_CLOCK_MODE_NONE: Publish nothing
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* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK: Publish the clock but not the offset
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* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET: Publish the clock and offset
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*
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* Whether the clock and possibly RTP/clock offset should be published according to RFC7273.
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*/
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typedef enum {
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GST_RTSP_PUBLISH_CLOCK_MODE_NONE,
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GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK,
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GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET
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} GstRTSPPublishClockMode;
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#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type())
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GST_RTSP_SERVER_API
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GType gst_rtsp_transport_mode_get_type (void);
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#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
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GST_RTSP_SERVER_API
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GType gst_rtsp_suspend_mode_get_type (void);
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#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type())
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GST_RTSP_SERVER_API
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GType gst_rtsp_publish_clock_mode_get_type (void);
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#include "rtsp-stream.h"
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#include "rtsp-thread-pool.h"
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#include "rtsp-permissions.h"
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#include "rtsp-address-pool.h"
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#include "rtsp-sdp.h"
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/**
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* GstRTSPMedia:
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*
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* A class that contains the GStreamer element along with a list of
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* #GstRTSPStream objects that can produce data.
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*
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* This object is usually created from a #GstRTSPMediaFactory.
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*/
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struct _GstRTSPMedia {
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GObject parent;
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/*< private >*/
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GstRTSPMediaPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstRTSPMediaClass:
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* @handle_message: handle a message
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* @prepare: the default implementation adds all elements and sets the
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* pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
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* in case of NO_PREROLL elements).
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* @unprepare: the default implementation sets the pipeline's state
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* to GST_STATE_NULL and removes all elements.
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* @suspend: the default implementation sets the pipeline's state to
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* GST_STATE_NULL GST_STATE_PAUSED depending on the selected
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* suspend mode.
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* @unsuspend: the default implementation reverts the suspend operation.
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* The pipeline will be prerolled again if it's state was
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* set to GST_STATE_NULL in suspend.
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* @convert_range: convert a range to the given unit
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* @query_position: query the current position in the pipeline
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* @query_stop: query when playback will stop
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*
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* The RTSP media class
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*/
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struct _GstRTSPMediaClass {
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GObjectClass parent_class;
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/* vmethods */
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gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
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gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
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gboolean (*unprepare) (GstRTSPMedia *media);
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gboolean (*suspend) (GstRTSPMedia *media);
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gboolean (*unsuspend) (GstRTSPMedia *media);
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gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
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GstRTSPRangeUnit unit);
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gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
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gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
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GstElement * (*create_rtpbin) (GstRTSPMedia *media);
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gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
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gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
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/* signals */
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void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
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void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
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void (*prepared) (GstRTSPMedia *media);
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void (*unprepared) (GstRTSPMedia *media);
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void (*target_state) (GstRTSPMedia *media, GstState state);
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void (*new_state) (GstRTSPMedia *media, GstState state);
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gboolean (*handle_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE-1];
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};
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GST_RTSP_SERVER_API
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GType gst_rtsp_media_get_type (void);
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/* creating the media */
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GST_RTSP_SERVER_API
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GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
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GST_RTSP_SERVER_API
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GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
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GST_RTSP_SERVER_API
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GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
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GstRTSPPermissions *permissions);
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GST_RTSP_SERVER_API
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GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
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GST_RTSP_SERVER_API
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GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
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GST_RTSP_SERVER_API
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GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
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GST_RTSP_SERVER_API
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GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
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GST_RTSP_SERVER_API
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GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface);
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GST_RTSP_SERVER_API
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gchar * gst_rtsp_media_get_multicast_iface (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
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GST_RTSP_SERVER_API
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guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
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GST_RTSP_SERVER_API
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GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
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gboolean do_retransmission);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
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GST_RTSP_SERVER_API
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guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
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const gchar *address, guint16 port);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode);
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GST_RTSP_SERVER_API
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GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl);
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GST_RTSP_SERVER_API
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guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia *media, gboolean bind_mcast_addr);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos);
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GST_RTSP_SERVER_API
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gint gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media);
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/* prepare the media for playback */
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
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GST_RTSP_SERVER_API
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GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
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GstSDPInfo * info);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
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/* creating streams */
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GST_RTSP_SERVER_API
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void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
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GstElement *payloader,
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GstPad *pad);
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/* dealing with the media */
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GST_RTSP_SERVER_API
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void gst_rtsp_media_lock (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_unlock (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
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GST_RTSP_SERVER_API
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GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media,
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GstRTSPTimeRange *range,
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GstSeekFlags flags);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_seek_trickmode (GstRTSPMedia *media,
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GstRTSPTimeRange *range,
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GstSeekFlags flags,
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gdouble rate,
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GstClockTime trickmode_interval);
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GST_RTSP_SERVER_API
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GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media);
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GST_RTSP_SERVER_API
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gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
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gboolean play,
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GstRTSPRangeUnit unit);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_get_rates (GstRTSPMedia * media,
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gdouble * rate,
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gdouble * applied_rate);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
|
|
GPtrArray *transports);
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GST_RTSP_SERVER_API
|
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void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
|
|
GstState state);
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|
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GST_RTSP_SERVER_API
|
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gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);
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|
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|
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
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref)
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#endif
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|
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|
G_END_DECLS
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#endif /* __GST_RTSP_MEDIA_H__ */
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