mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-01 14:11:15 +00:00
3aa69ca0bb
This is needed in order to: - Avoid ignoring requests for different media sources. - Add SSRC field in the GstForceKeyUnit event. https://bugzilla.gnome.org/show_bug.cgi?id=778013
403 lines
16 KiB
C
403 lines
16 KiB
C
/* 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
|
|
* @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
|
|
* @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);
|
|
|
|
/**
|
|
* 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;
|
|
|
|
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__ */
|