mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-12 03:16:33 +00:00
f52e16ceb8
This reverts commit dcd3ce9751
.
This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.
This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.
Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.
Fixes #537
144 lines
5.4 KiB
C
144 lines
5.4 KiB
C
/* GStreamer
|
|
* Copyright (C) <2007> Wim Taymans <wim.taymans@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.
|
|
*/
|
|
|
|
#ifndef __GST_RTP_BIN_H__
|
|
#define __GST_RTP_BIN_H__
|
|
|
|
#include <gst/gst.h>
|
|
|
|
#include "rtpsession.h"
|
|
#include "gstrtpsession.h"
|
|
#include "rtpjitterbuffer.h"
|
|
|
|
#define GST_TYPE_RTP_BIN \
|
|
(gst_rtp_bin_get_type())
|
|
#define GST_RTP_BIN(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BIN,GstRtpBin))
|
|
#define GST_RTP_BIN_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BIN,GstRtpBinClass))
|
|
#define GST_IS_RTP_BIN(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BIN))
|
|
#define GST_IS_RTP_BIN_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BIN))
|
|
|
|
typedef enum
|
|
{
|
|
GST_RTP_BIN_RTCP_SYNC_ALWAYS,
|
|
GST_RTP_BIN_RTCP_SYNC_INITIAL,
|
|
GST_RTP_BIN_RTCP_SYNC_RTP
|
|
} GstRTCPSync;
|
|
|
|
typedef struct _GstRtpBin GstRtpBin;
|
|
typedef struct _GstRtpBinClass GstRtpBinClass;
|
|
typedef struct _GstRtpBinPrivate GstRtpBinPrivate;
|
|
|
|
struct _GstRtpBin {
|
|
GstBin bin;
|
|
|
|
/*< private >*/
|
|
/* default latency for sessions */
|
|
guint latency_ms;
|
|
guint64 latency_ns;
|
|
gboolean drop_on_latency;
|
|
gboolean do_lost;
|
|
gboolean ignore_pt;
|
|
gboolean ntp_sync;
|
|
gint rtcp_sync;
|
|
guint rtcp_sync_interval;
|
|
RTPJitterBufferMode buffer_mode;
|
|
gboolean buffering;
|
|
gboolean use_pipeline_clock;
|
|
GstRtpNtpTimeSource ntp_time_source;
|
|
gboolean send_sync_event;
|
|
GstClockTime buffer_start;
|
|
gboolean do_retransmission;
|
|
GstRTPProfile rtp_profile;
|
|
gboolean rtcp_sync_send_time;
|
|
gint max_rtcp_rtp_time_diff;
|
|
guint32 max_dropout_time;
|
|
guint32 max_misorder_time;
|
|
gboolean rfc7273_sync;
|
|
guint max_streams;
|
|
guint64 max_ts_offset_adjustment;
|
|
gint64 max_ts_offset;
|
|
gboolean max_ts_offset_is_set;
|
|
|
|
/* a list of session */
|
|
GSList *sessions;
|
|
|
|
/* a list of clients, these are streams with the same CNAME */
|
|
GSList *clients;
|
|
|
|
/* the default SDES items for sessions */
|
|
GstStructure *sdes;
|
|
|
|
/*< private >*/
|
|
GstRtpBinPrivate *priv;
|
|
};
|
|
|
|
struct _GstRtpBinClass {
|
|
GstBinClass parent_class;
|
|
|
|
/* get the caps for pt */
|
|
GstCaps* (*request_pt_map) (GstRtpBin *rtpbin, guint session, guint pt);
|
|
|
|
void (*payload_type_change) (GstRtpBin *rtpbin, guint session, guint pt);
|
|
|
|
void (*new_jitterbuffer) (GstRtpBin *rtpbin, GstElement *jitterbuffer, guint session, guint32 ssrc);
|
|
|
|
void (*new_storage) (GstRtpBin *rtpbin, GstElement *jitterbuffer, guint session);
|
|
|
|
/* action signals */
|
|
void (*clear_pt_map) (GstRtpBin *rtpbin);
|
|
void (*reset_sync) (GstRtpBin *rtpbin);
|
|
GstElement* (*get_session) (GstRtpBin *rtpbin, guint session);
|
|
RTPSession* (*get_internal_session) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*get_storage) (GstRtpBin *rtpbin, guint session);
|
|
GObject* (*get_internal_storage) (GstRtpBin *rtpbin, guint session);
|
|
|
|
/* session manager signals */
|
|
void (*on_new_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_ssrc_collision) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_ssrc_validated) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_ssrc_active) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_ssrc_sdes) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_bye_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_bye_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_sender_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_npt_stop) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
|
|
GstElement* (*request_rtp_encoder) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*request_rtp_decoder) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*request_rtcp_encoder) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*request_rtcp_decoder) (GstRtpBin *rtpbin, guint session);
|
|
|
|
GstElement* (*request_aux_sender) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*request_aux_receiver) (GstRtpBin *rtpbin, guint session);
|
|
|
|
GstElement* (*request_fec_encoder) (GstRtpBin *rtpbin, guint session);
|
|
GstElement* (*request_fec_decoder) (GstRtpBin *rtpbin, guint session);
|
|
|
|
void (*on_new_sender_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
void (*on_sender_ssrc_active) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
|
|
};
|
|
|
|
GType gst_rtp_bin_get_type (void);
|
|
|
|
#endif /* __GST_RTP_BIN_H__ */
|