/* 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 * 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 __RTP_SESSION_H__ #define __RTP_SESSION_H__ #include #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, gpointer data, gpointer user_data); /** * RTPSessionSendRTCP: * @sess: an #RTPSession * @src: the #RTPSource * @buffer: the RTCP 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 (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, 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 * @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, 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); /** * 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 * * 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; } 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; RTPSessionStats stats; RTPSessionStats bye_stats; gboolean favor_new; GstClockTime rtcp_feedback_retention_window; guint rtcp_immediate_feedback_threshold; GstClockTime last_keyframe_request; gboolean last_keyframe_all_headers; 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__ */