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/* GStreamer
* Copyright (C) <2007> Wim Taymans <>
* 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
* 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 __RTP_SESSION_H__
#define __RTP_SESSION_H__
#include <gst/gst.h>
#include "rtpsource.h"
typedef struct _RTPSession RTPSession;
typedef struct _RTPSessionClass RTPSessionClass;
#define RTP_TYPE_SESSION (rtp_session_get_type())
#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, gpointer data, 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
* @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, 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);
* RTPSessionRequestKeyUnit:
* @sess: an #RTPSession
* @ssrc: SSRC of the source related to the key unit request
* @all_headers: %TRUE if "all-headers" property should be set on the key unit
* request
* @user_data: user data specified when registering
* Asks the encoder to produce a key unit as soon as possibly within the
* bandwidth constraints
typedef void (*RTPSessionRequestKeyUnit) (RTPSession *sess, guint32 ssrc,
gboolean all_headers, gpointer user_data);
* RTPSessionRequestTime:
* @sess: an #RTPSession
* @user_data: user data specified when registering
* This callback will be called when @sess needs the current time. The time
* should be returned as a #GstClockTime
typedef GstClockTime (*RTPSessionRequestTime) (RTPSession *sess,
gpointer user_data);
* RTPSessionNotifyNACK:
* @sess: an #RTPSession
* @seqnum: the missing seqnum
* @blp: other missing seqnums
* @ssrc: SSRC of requested stream
* @user_data: user data specified when registering
* Notifies of NACKed frames.
typedef void (*RTPSessionNotifyNACK) (RTPSession *sess,
guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data);
* RTPSessionReconfigure:
* @sess: an #RTPSession
* @user_data: user data specified when registering
* This callback will be called when @sess wants to reconfigure the
* negotiated parameters.
typedef void (*RTPSessionReconfigure) (RTPSession *sess, gpointer user_data);
* RTPSessionNotifyEarlyRTCP:
* @sess: an #RTPSession
* @user_data: user data specified when registering
* Notifies of early RTCP being requested
typedef void (*RTPSessionNotifyEarlyRTCP) (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
* @RTPSessionRequestKeyUnit: callback for requesting a new key unit
* @RTPSessionRequestTime: callback for requesting the current time
* @RTPSessionNotifyNACK: callback for notifying NACK
* @RTPSessionReconfigure: callback for requesting reconfiguration
* @RTPSessionNotifyEarlyRTCP: callback for notifying early RTCP
* 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;
RTPSessionRequestKeyUnit request_key_unit;
RTPSessionRequestTime request_time;
RTPSessionNotifyNACK notify_nack;
RTPSessionReconfigure reconfigure;
RTPSessionNotifyEarlyRTCP notify_early_rtcp;
} RTPSessionCallbacks;
* RTPSession:
* @lock: lock to protect the session
* @source: the source of this session
* @ssrcs: Hashtable of sources indexed by SSRC
* @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;
GstStructure *sdes;
guint probation;
guint32 max_dropout_time;
guint32 max_misorder_time;
GstRTPProfile rtp_profile;
gboolean reduced_size_rtcp;
/* bandwidths */
gboolean recalc_bandwidth;
guint bandwidth;
gdouble rtcp_bandwidth;
guint rtcp_rr_bandwidth;
guint rtcp_rs_bandwidth;
guint32 suggested_ssrc;
gboolean internal_ssrc_set;
gboolean internal_ssrc_from_caps_or_property;
/* for sender/receiver counting */
guint32 key;
guint32 mask_idx;
guint32 mask;
GHashTable *ssrcs[32];
guint total_sources;
guint16 generation;
GstClockTime next_rtcp_check_time; /* tn */
GstClockTime last_rtcp_check_time; /* tp */
GstClockTime last_rtcp_send_time; /* t_rr_last */
GstClockTime last_rtcp_interval; /* T_rr */
GstClockTime start_time;
gboolean first_rtcp;
gboolean allow_early;
GstClockTime next_early_rtcp_time;
gboolean scheduled_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;
gpointer request_key_unit_user_data;
gpointer request_time_user_data;
gpointer notify_nack_user_data;
gpointer reconfigure_user_data;
gpointer notify_early_rtcp_user_data;
RTPSessionStats stats;
RTPSessionStats bye_stats;
gboolean favor_new;
GstClockTime rtcp_feedback_retention_window;
guint rtcp_immediate_feedback_threshold;
gboolean is_doing_ptp;
GList *conflicting_addresses;
* 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;
/* action signals */
RTPSource* (*get_source_by_ssrc) (RTPSession *sess, guint32 ssrc);
/* 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);
gboolean (*on_sending_rtcp) (RTPSession *sess, GstBuffer *buffer,
gboolean early);
void (*on_app_rtcp) (RTPSession *sess, guint subtype, guint ssrc,
const gchar *name, GstBuffer *data);
void (*on_feedback_rtcp) (RTPSession *sess, guint type, guint fbtype,
guint sender_ssrc, guint media_ssrc, GstBuffer *fci);
gboolean (*send_rtcp) (RTPSession *sess, GstClockTime max_delay);
void (*on_receiving_rtcp) (RTPSession *sess, GstBuffer *buffer);
void (*on_new_sender_ssrc) (RTPSession *sess, RTPSource *source);
void (*on_sender_ssrc_active) (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_request_time_callback (RTPSession * sess,
RTPSessionRequestTime 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);
GstStructure * rtp_session_get_sdes_struct (RTPSession *sess);
void rtp_session_set_sdes_struct (RTPSession *sess, const GstStructure *sdes);
/* handling sources */
guint32 rtp_session_suggest_ssrc (RTPSession *sess, gboolean *is_random);
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_create_source (RTPSession *sess);
/* processing packets from receivers */
GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer,
GstClockTime current_time,
GstClockTime running_time,
guint64 ntpnstime);
GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer,
GstClockTime current_time,
guint64 ntpnstime);
/* processing packets for sending */
void rtp_session_update_send_caps (RTPSession *sess, GstCaps *caps);
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list,
GstClockTime current_time, GstClockTime running_time);
/* scheduling bye */
void rtp_session_mark_all_bye (RTPSession *sess, const gchar *reason);
GstFlowReturn rtp_session_schedule_bye (RTPSession *sess, 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, GstClockTime running_time);
/* request the transmittion of an early RTCP packet */
gboolean rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
GstClockTime max_delay);
/* Notify session of a request for a new key unit */
gboolean rtp_session_request_key_unit (RTPSession * sess,
guint32 ssrc,
gboolean fir,
gint count);
gboolean rtp_session_request_nack (RTPSession * sess,
guint32 ssrc,
guint16 seqnum,
GstClockTime max_delay);
#endif /* __RTP_SESSION_H__ */