mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 19:06:33 +00:00
7cfd59820a
When the underlying layers are running on_message_sent, this sometimes causes the underlying layer to send more data, which will cause the underlying layer to run callback on_message_sent again. This can go on and on. To break this chain, we introduce an idle source that takes care of sending data if there are more to send when running callback https://bugzilla.gnome.org/show_bug.cgi?id=797289
395 lines
15 KiB
C
395 lines
15 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 <gio/gio.h>
|
|
|
|
#ifndef __GST_RTSP_STREAM_H__
|
|
#define __GST_RTSP_STREAM_H__
|
|
|
|
#include "rtsp-server-prelude.h"
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
/* types for the media stream */
|
|
#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
|
|
#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
|
|
#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
|
|
#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
|
|
#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
|
|
#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
|
|
#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
|
|
#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
|
|
|
|
typedef struct _GstRTSPStream GstRTSPStream;
|
|
typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
|
|
typedef struct _GstRTSPStreamPrivate GstRTSPStreamPrivate;
|
|
|
|
#include "rtsp-stream-transport.h"
|
|
#include "rtsp-address-pool.h"
|
|
#include "rtsp-session.h"
|
|
#include "rtsp-media.h"
|
|
|
|
/**
|
|
* GstRTSPStream:
|
|
*
|
|
* The definition of a media stream.
|
|
*/
|
|
struct _GstRTSPStream {
|
|
GObject parent;
|
|
|
|
/*< private >*/
|
|
GstRTSPStreamPrivate *priv;
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
struct _GstRTSPStreamClass {
|
|
GObjectClass parent_class;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GST_RTSP_SERVER_API
|
|
GType gst_rtsp_stream_get_type (void);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
|
|
GstPad *pad);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_index (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_pt (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstPad * gst_rtsp_stream_get_srcpad (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstPad * gst_rtsp_stream_get_sinkpad (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_control (GstRTSPStream *stream, const gchar *control);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gchar * gst_rtsp_stream_get_control (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_has_control (GstRTSPStream *stream, const gchar *control);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_mtu (GstRTSPStream *stream, guint mtu);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_mtu (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream, gint dscp_qos);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gint gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_transport_supported (GstRTSPStream *stream,
|
|
GstRTSPTransport *transport);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_profiles (GstRTSPStream *stream, GstRTSPProfile profiles);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPProfile gst_rtsp_stream_get_profiles (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_protocols (GstRTSPStream *stream, GstRTSPLowerTrans protocols);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPLowerTrans gst_rtsp_stream_get_protocols (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPAddressPool *
|
|
gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_multicast_iface (GstRTSPStream *stream, const gchar * multicast_iface);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gchar * gst_rtsp_stream_get_multicast_iface (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPAddress * gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
|
|
const gchar * address,
|
|
guint port,
|
|
guint n_ports,
|
|
guint ttl);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_join_bin (GstRTSPStream *stream,
|
|
GstBin *bin, GstElement *rtpbin,
|
|
GstState state);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
|
|
GstBin *bin, GstElement *rtpbin);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstBin * gst_rtsp_stream_get_joined_bin (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_set_blocked (GstRTSPStream * stream,
|
|
gboolean blocked);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_blocking (GstRTSPStream * stream);
|
|
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_unblock_linked (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_client_side (GstRTSPStream *stream, gboolean client_side);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_client_side (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
|
|
GstRTSPRange *server_port,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPAddress * gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
|
|
GSocketFamily family);
|
|
|
|
|
|
GST_RTSP_SERVER_API
|
|
GObject * gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstElement * gst_rtsp_stream_get_srtp_encoder (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
|
|
guint *ssrc);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
|
|
guint *rtptime, guint *seq,
|
|
guint *clock_rate,
|
|
GstClockTime *running_time);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstCaps * gst_rtsp_stream_get_caps (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
|
|
GstBuffer *buffer);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
|
|
GstBuffer *buffer);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
|
|
GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
|
|
GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GSocket * gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GSocket * gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GSocket * gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream *stream,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GSocket * gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream *stream,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
|
|
const gchar * destination,
|
|
guint rtp_port,
|
|
guint rtcp_port,
|
|
GSocketFamily family);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gchar * gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
|
|
guint ssrc, GstCaps * crypto);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_query_position (GstRTSPStream * stream,
|
|
gint64 * position);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_query_stop (GstRTSPStream * stream,
|
|
gint64 * stop);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_seekable (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_seqnum_offset (GstRTSPStream *stream, guint16 seqnum);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint16 gst_rtsp_stream_get_current_seqnum (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_retransmission_time (GstRTSPStream *stream, GstClockTime time);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstClockTime gst_rtsp_stream_get_retransmission_time (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream,
|
|
guint rtx_pt);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_buffer_size (GstRTSPStream *stream, guint size);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_buffer_size (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstElement * gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstElement * gst_rtsp_stream_request_aux_receiver (GstRTSPStream * stream, guint sessid);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream, GSocketFamily family,
|
|
GstRTSPTransport *transport, gboolean use_client_settings);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream, GstRTSPPublishClockMode mode);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPPublishClockMode gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream *stream, guint ttl);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream *stream, guint ttl);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream, gboolean bind_mcast_addr);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_complete_stream (GstRTSPStream * stream, const GstRTSPTransport * transport);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_complete (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_sender (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_is_receiver (GstRTSPStream * stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_handle_keymgmt (GstRTSPStream *stream, const gchar *keymgmt);
|
|
|
|
/* ULP Forward Error Correction (RFC 5109) */
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_get_ulpfec_enabled (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_ulpfec_pt (GstRTSPStream *stream, guint pt);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_ulpfec_pt (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstElement * gst_rtsp_stream_request_ulpfec_decoder (GstRTSPStream *stream, GstElement *rtpbin, guint sessid);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstElement * gst_rtsp_stream_request_ulpfec_encoder (GstRTSPStream *stream, guint sessid);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_ulpfec_percentage (GstRTSPStream *stream, guint percentage);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_stream_get_ulpfec_percentage (GstRTSPStream *stream);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_set_watch_context (GstRTSPStream * stream, GMainContext * context);
|
|
|
|
/**
|
|
* GstRTSPStreamTransportFilterFunc:
|
|
* @stream: a #GstRTSPStream object
|
|
* @trans: a #GstRTSPStreamTransport in @stream
|
|
* @user_data: user data that has been given to gst_rtsp_stream_transport_filter()
|
|
*
|
|
* This function will be called by the gst_rtsp_stream_transport_filter(). An
|
|
* implementation should return a value of #GstRTSPFilterResult.
|
|
*
|
|
* When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed
|
|
* from @stream.
|
|
*
|
|
* A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in
|
|
* @stream.
|
|
*
|
|
* A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of
|
|
* gst_rtsp_stream_transport_filter().
|
|
*
|
|
* Returns: a #GstRTSPFilterResult.
|
|
*/
|
|
typedef GstRTSPFilterResult (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
|
|
GstRTSPStreamTransport *trans,
|
|
gpointer user_data);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GList * gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
|
|
GstRTSPStreamTransportFilterFunc func,
|
|
gpointer user_data);
|
|
|
|
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStream, gst_object_unref)
|
|
#endif
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_STREAM_H__ */
|