gstreamer/gst/rtpmanager/gstrtpsession.h
Wim Taymans a35d1dde42 gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
2008-09-05 13:52:34 +00:00

81 lines
2.9 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_RTP_SESSION_H__
#define __GST_RTP_SESSION_H__
#include <gst/gst.h>
#define GST_TYPE_RTP_SESSION \
(gst_rtp_session_get_type())
#define GST_RTP_SESSION(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SESSION,GstRtpSession))
#define GST_RTP_SESSION_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SESSION,GstRtpSessionClass))
#define GST_IS_RTP_SESSION(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SESSION))
#define GST_IS_RTP_SESSION_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SESSION))
#define GST_RTP_SESSION_CAST(obj) ((GstRtpSession *)(obj))
typedef struct _GstRtpSession GstRtpSession;
typedef struct _GstRtpSessionClass GstRtpSessionClass;
typedef struct _GstRtpSessionPrivate GstRtpSessionPrivate;
struct _GstRtpSession {
GstElement element;
/*< private >*/
GstPad *recv_rtp_sink;
GstSegment recv_rtp_seg;
GstPad *recv_rtcp_sink;
GstPad *send_rtp_sink;
GstSegment send_rtp_seg;
GstPad *recv_rtp_src;
GstPad *sync_src;
GstPad *send_rtp_src;
GstPad *send_rtcp_src;
GstRtpSessionPrivate *priv;
};
struct _GstRtpSessionClass {
GstElementClass parent_class;
/* signals */
GstCaps* (*request_pt_map) (GstRtpSession *sess, guint pt);
void (*clear_pt_map) (GstRtpSession *sess);
void (*on_new_ssrc) (GstRtpSession *sess, guint32 ssrc);
void (*on_ssrc_collision) (GstRtpSession *sess, guint32 ssrc);
void (*on_ssrc_validated) (GstRtpSession *sess, guint32 ssrc);
void (*on_ssrc_active) (GstRtpSession *sess, guint32 ssrc);
void (*on_ssrc_sdes) (GstRtpSession *sess, guint32 ssrc);
void (*on_bye_ssrc) (GstRtpSession *sess, guint32 ssrc);
void (*on_bye_timeout) (GstRtpSession *sess, guint32 ssrc);
void (*on_timeout) (GstRtpSession *sess, guint32 ssrc);
void (*on_sender_timeout) (GstRtpSession *sess, guint32 ssrc);
};
GType gst_rtp_session_get_type (void);
void gst_rtp_session_set_ssrc (GstRtpSession *sess, guint32 ssrc);
#endif /* __GST_RTP_SESSION_H__ */