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85e26f6546
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.
82 lines
2.9 KiB
C
82 lines
2.9 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_RTP_SESSION_H__
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#define __GST_RTP_SESSION_H__
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#include <gst/gst.h>
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#define GST_TYPE_RTP_SESSION \
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(gst_rtp_session_get_type())
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#define GST_RTP_SESSION(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SESSION,GstRtpSession))
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#define GST_RTP_SESSION_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SESSION,GstRtpSessionClass))
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#define GST_IS_RTP_SESSION(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SESSION))
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#define GST_IS_RTP_SESSION_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SESSION))
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#define GST_RTP_SESSION_CAST(obj) ((GstRtpSession *)(obj))
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typedef struct _GstRtpSession GstRtpSession;
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typedef struct _GstRtpSessionClass GstRtpSessionClass;
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typedef struct _GstRtpSessionPrivate GstRtpSessionPrivate;
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struct _GstRtpSession {
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GstElement element;
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/*< private >*/
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GstPad *recv_rtp_sink;
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GstSegment recv_rtp_seg;
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GstPad *recv_rtcp_sink;
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GstPad *send_rtp_sink;
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GstSegment send_rtp_seg;
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GstPad *recv_rtp_src;
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GstPad *sync_src;
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GstPad *send_rtp_src;
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GstPad *send_rtcp_src;
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GstRtpSessionPrivate *priv;
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};
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struct _GstRtpSessionClass {
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GstElementClass parent_class;
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/* signals */
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GstCaps* (*request_pt_map) (GstRtpSession *sess, guint pt);
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void (*clear_pt_map) (GstRtpSession *sess);
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void (*on_new_ssrc) (GstRtpSession *sess, guint32 ssrc);
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void (*on_ssrc_collision) (GstRtpSession *sess, guint32 ssrc);
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void (*on_ssrc_validated) (GstRtpSession *sess, guint32 ssrc);
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void (*on_ssrc_active) (GstRtpSession *sess, guint32 ssrc);
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void (*on_ssrc_sdes) (GstRtpSession *sess, guint32 ssrc);
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void (*on_bye_ssrc) (GstRtpSession *sess, guint32 ssrc);
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void (*on_bye_timeout) (GstRtpSession *sess, guint32 ssrc);
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void (*on_timeout) (GstRtpSession *sess, guint32 ssrc);
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void (*on_sender_timeout) (GstRtpSession *sess, guint32 ssrc);
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};
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GType gst_rtp_session_get_type (void);
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void gst_rtp_session_set_ssrc (GstRtpSession *sess, guint32 ssrc);
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#endif /* __GST_RTP_SESSION_H__ */
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