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gstreamer/gst/rtpmanager/rtpsession.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

295 lines
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/* 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 __RTP_SESSION_H__
#define __RTP_SESSION_H__
#include <gst/gst.h>
#include <gst/netbuffer/gstnetbuffer.h>
#include "rtpsource.h"
typedef struct _RTPSession RTPSession;
typedef struct _RTPSessionClass RTPSessionClass;
#define RTP_TYPE_SESSION (rtp_session_get_type())
#define RTP_SESSION(sess) (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession))
#define RTP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass))
#define RTP_IS_SESSION(sess) (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION))
#define RTP_IS_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION))
#define RTP_SESSION_CAST(sess) ((RTPSession *)(sess))
#define RTP_SESSION_LOCK(sess) (g_mutex_lock ((sess)->lock))
#define RTP_SESSION_UNLOCK(sess) (g_mutex_unlock ((sess)->lock))
/**
* RTPSessionProcessRTP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTP buffer ready for processing
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for further
* processing. Processing the buffer typically includes decoding and displaying
* the buffer.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionSendRTP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTP buffer ready for sending
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for sending to
* all listening participants in this session.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionSendRTCP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTCP buffer ready for sending
* @eos: if an EOS event should be pushed
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for sending to
* all listening participants in this session.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer,
gboolean eos, gpointer user_data);
/**
* RTPSessionSyncRTCP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTCP buffer ready for synchronisation
* @user_data: user data specified when registering
*
* This callback will be called when @sess has an SR @buffer ready for doing
* synchronisation between streams.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSyncRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionClockRate:
* @sess: an #RTPSession
* @payload: the payload
* @user_data: user data specified when registering
*
* This callback will be called when @sess needs the clock-rate of @payload.
*
* Returns: the clock-rate of @pt.
*/
typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
/**
* RTPSessionReconsider:
* @sess: an #RTPSession
* @user_data: user data specified when registering
*
* This callback will be called when @sess needs to cancel the current timeout.
* The currently running timeout should be canceled and a new reporting interval
* should be requested from @sess.
*/
typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
/**
* RTPSessionCallbacks:
* @RTPSessionProcessRTP: callback to process RTP packets
* @RTPSessionSendRTP: callback for sending RTP packets
* @RTPSessionSendRTCP: callback for sending RTCP packets
* @RTPSessionSyncRTCP: callback for handling SR packets
* @RTPSessionReconsider: callback for reconsidering the timeout
*
* These callbacks can be installed on the session manager to get notification
* when RTP and RTCP packets are ready for further processing. These callbacks
* are not implemented with signals for performance reasons.
*/
typedef struct {
RTPSessionProcessRTP process_rtp;
RTPSessionSendRTP send_rtp;
RTPSessionSyncRTCP sync_rtcp;
RTPSessionSendRTCP send_rtcp;
RTPSessionClockRate clock_rate;
RTPSessionReconsider reconsider;
} RTPSessionCallbacks;
/**
* RTPConflictingAddress:
* @address: #GstNetAddress which conflicted
* @last_conflict_time: time when the last conflict was seen
*
* This structure is used to account for addresses that have conflicted to find
* loops.
*/
typedef struct {
GstNetAddress address;
GstClockTime time;
} RTPConflictingAddress;
/**
* RTPSession:
* @lock: lock to protect the session
* @source: the source of this session
* @ssrcs: Hashtable of sources indexed by SSRC
* @cnames: Hashtable of sources indexed by CNAME
* @num_sources: the number of sources
* @activecount: the number of active sources
* @callbacks: callbacks
* @user_data: user data passed in callbacks
* @stats: session statistics
* @conflicting_addresses: GList of conflicting addresses
*
* The RTP session manager object
*/
struct _RTPSession {
GObject object;
GMutex *lock;
guint header_len;
guint mtu;
RTPSource *source;
/* for sender/receiver counting */
guint32 key;
guint32 mask_idx;
guint32 mask;
GHashTable *ssrcs[32];
GHashTable *cnames;
guint total_sources;
GstClockTime next_rtcp_check_time;
GstClockTime last_rtcp_send_time;
gboolean first_rtcp;
gchar *bye_reason;
gboolean sent_bye;
RTPSessionCallbacks callbacks;
gpointer process_rtp_user_data;
gpointer send_rtp_user_data;
gpointer send_rtcp_user_data;
gpointer sync_rtcp_user_data;
gpointer clock_rate_user_data;
gpointer reconsider_user_data;
RTPSessionStats stats;
GList *conflicting_addresses;
gboolean change_ssrc;
};
/**
* RTPSessionClass:
* @on_new_ssrc: emited when a new source is found
* @on_bye_ssrc: emited when a source is gone
*
* The session class.
*/
struct _RTPSessionClass {
GObjectClass parent_class;
/* signals */
void (*on_new_ssrc) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_active) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_sdes) (RTPSession *sess, RTPSource *source);
void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source);
void (*on_bye_timeout) (RTPSession *sess, RTPSource *source);
void (*on_timeout) (RTPSession *sess, RTPSource *source);
void (*on_sender_timeout) (RTPSession *sess, RTPSource *source);
};
GType rtp_session_get_type (void);
/* create and configure */
RTPSession* rtp_session_new (void);
void rtp_session_set_callbacks (RTPSession *sess,
RTPSessionCallbacks *callbacks,
gpointer user_data);
void rtp_session_set_process_rtp_callback (RTPSession * sess,
RTPSessionProcessRTP callback,
gpointer user_data);
void rtp_session_set_send_rtp_callback (RTPSession * sess,
RTPSessionSendRTP callback,
gpointer user_data);
void rtp_session_set_send_rtcp_callback (RTPSession * sess,
RTPSessionSendRTCP callback,
gpointer user_data);
void rtp_session_set_sync_rtcp_callback (RTPSession * sess,
RTPSessionSyncRTCP callback,
gpointer user_data);
void rtp_session_set_clock_rate_callback (RTPSession * sess,
RTPSessionClockRate callback,
gpointer user_data);
void rtp_session_set_reconsider_callback (RTPSession * sess,
RTPSessionReconsider callback,
gpointer user_data);
void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth);
gdouble rtp_session_get_bandwidth (RTPSession *sess);
void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
gboolean rtp_session_set_sdes_string (RTPSession *sess, GstRTCPSDESType type,
const gchar *cname);
gchar* rtp_session_get_sdes_string (RTPSession *sess, GstRTCPSDESType type);
/* handling sources */
RTPSource* rtp_session_get_internal_source (RTPSession *sess);
void rtp_session_set_internal_ssrc (RTPSession *sess, guint32 ssrc);
guint32 rtp_session_get_internal_ssrc (RTPSession *sess);
gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
guint rtp_session_get_num_sources (RTPSession *sess);
guint rtp_session_get_num_active_sources (RTPSession *sess);
RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc);
RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname);
RTPSource* rtp_session_create_source (RTPSession *sess);
/* processing packets from receivers */
GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer, GstClockTime current_time, guint64 ntpnstime);
GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer, GstClockTime current_time);
/* processing packets for sending */
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer, GstClockTime current_time, guint64 ntpnstime);
/* stopping the session */
GstFlowReturn rtp_session_send_bye (RTPSession *sess, const gchar *reason, GstClockTime current_time);
/* get interval for next RTCP interval */
GstClockTime rtp_session_next_timeout (RTPSession *sess, GstClockTime current_time);
GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTime current_time, guint64 ntpnstime);
#endif /* __RTP_SESSION_H__ */
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