| /* 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., 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(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 |
| * @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__ */ |
