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Original commit message from CVS: * configure.ac: Disable rtpmanager for now because it depends on CVS -base. * gst/rtpmanager/Makefile.am: Added new files for session manager. * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (pt_map_requested), (new_ssrc_pad_found): Some cleanups. the session manager can now also request a pt-map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate), (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpsession.h: We can ask for pt-map now too when the session manager needs it. Hook up to the new session manager, implement the needed callbacks for pushing data, getting clock time and requesting clock-rates. Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to be send to clients. Add code to start and stop the thread that will schedule RTCP through the session manager. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth), (source_push_rtp), (source_clock_rate), (check_collision), (obtain_source), (rtp_session_add_source), (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_get_source_by_ssrc), (rtp_session_get_source_by_cname), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_process_app), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_rtcp_interval), (rtp_session_produce_rtcp): * gst/rtpmanager/rtpsession.h: The advanced beginnings of the main session manager that handles the participant database of RTPSources, SSRC probation, SSRC collisions, parse RTCP to update source stats. etc.. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_finalize), (rtp_source_new), (rtp_source_set_callbacks), (rtp_source_set_as_csrc), (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from), (push_packet), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_process_bye), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb): * gst/rtpmanager/rtpsource.h: Object that encapsulates an SSRC and its state in the database. Calculates the jitter and transit times of data packets. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter): * gst/rtpmanager/rtpstats.h: Various stats regarding the session and sources. Used to calculate the RTCP interval.
207 lines
7.2 KiB
C
207 lines
7.2 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __RTP_SESSION_H__
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#define __RTP_SESSION_H__
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#include <gst/gst.h>
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#include <gst/netbuffer/gstnetbuffer.h>
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#include "rtpsource.h"
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typedef struct _RTPSession RTPSession;
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typedef struct _RTPSessionClass RTPSessionClass;
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#define RTP_TYPE_SESSION (rtp_session_get_type())
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#define RTP_SESSION(sess) (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession))
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#define RTP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass))
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#define RTP_IS_SESSION(sess) (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION))
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#define RTP_IS_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION))
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#define RTP_SESSION_CAST(sess) ((RTPSession *)(sess))
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#define RTP_SESSION_LOCK(sess) (g_mutex_lock ((sess)->lock))
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#define RTP_SESSION_UNLOCK(sess) (g_mutex_unlock ((sess)->lock))
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/**
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* RTPSessionProcessRTP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @buffer: the RTP buffer ready for processing
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* @user_data: user data specified when registering
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*
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* This callback will be called when @sess has @buffer ready for further
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* processing. Processing the buffer typically includes decoding and displaying
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* the buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionSendRTP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @buffer: the RTP buffer ready for sending
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* @user_data: user data specified when registering
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*
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* This callback will be called when @sess has @buffer ready for sending to
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* all listening participants in this session.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionSendRTCP:
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* @sess: an #RTPSession
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* @src: the #RTPSource
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* @buffer: the RTCP buffer ready for sending
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* @user_data: user data specified when registering
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*
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* This callback will be called when @sess has @buffer ready for sending to
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* all listening participants in this session.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
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/**
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* RTPSessionClockRate:
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* @sess: an #RTPSession
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* @payload: the payload
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* @user_data: user data specified when registering
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*
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* This callback will be called when @sess needs the clock-rate of @payload.
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*
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* Returns: the clock-rate of @pt.
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*/
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typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
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/**
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* RTPSessionGetTime:
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* @sess: an #RTPSession
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* @user_data: user data specified when registering
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*
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* This callback will be called when @sess needs the current time in
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* nanoseconds.
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*
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* Returns: a #GstClockTime with the current time in nanoseconds.
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*/
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typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data);
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/**
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* RTPSessionCallbacks:
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* @RTPSessionProcessRTP: callback to process RTP packets
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* @RTPSessionSendRTP: callback for sending RTP packets
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* @RTPSessionSendRTCP: callback for sending RTCP packets
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* @RTPSessionGetTime: callback for returning the current time
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*
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* These callbacks can be installed on the session manager to get notification
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* when RTP and RTCP packets are ready for further processing. These callbacks
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* are not implemented with signals for performance reasons.
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*/
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typedef struct {
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RTPSessionProcessRTP process_rtp;
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RTPSessionSendRTP send_rtp;
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RTPSessionSendRTCP send_rtcp;
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RTPSessionClockRate clock_rate;
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RTPSessionGetTime get_time;
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} RTPSessionCallbacks;
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/**
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* RTPSession:
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* @lock: lock to protect the session
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* @source: the source of this session
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* @ssrcs: Hashtable of sources indexed by SSRC
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* @cnames: Hashtable of sources indexed by CNAME
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* @num_sources: the number of sources
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* @activecount: the number of active sources
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* @callbacks: callbacks
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* @user_data: user data passed in callbacks
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*
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* The RTP session manager object
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*/
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struct _RTPSession {
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GObject object;
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GMutex *lock;
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guint header_len;
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RTPSource *source;
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GHashTable *ssrcs;
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GHashTable *cnames;
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guint total_sources;
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RTPSessionCallbacks callbacks;
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gpointer user_data;
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RTPSessionStats stats;
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};
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/**
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* RTPSessionClass:
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* @on_new_ssrc: emited when a new source is found
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* @on_bye_ssrc: emited when a source is gone
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*
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* The session class.
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*/
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struct _RTPSessionClass {
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GObjectClass parent_class;
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/* signals */
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void (*on_new_ssrc) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source);
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void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source);
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void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source);
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};
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GType rtp_session_get_type (void);
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/* create and configure */
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RTPSession* rtp_session_new (void);
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void rtp_session_set_callbacks (RTPSession *sess,
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RTPSessionCallbacks *callbacks,
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gpointer user_data);
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void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth);
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gdouble rtp_session_get_bandwidth (RTPSession *sess);
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void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
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gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
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/* handling sources */
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gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
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gint rtp_session_get_num_sources (RTPSession *sess);
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gint rtp_session_get_num_active_sources (RTPSession *sess);
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RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc);
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RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname);
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RTPSource* rtp_session_create_source (RTPSession *sess);
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/* processing packets from receivers */
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GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer);
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GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer);
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/* processing packets for sending */
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GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer);
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/* get interval for next RTCP interval */
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gdouble rtp_session_get_rtcp_interval (RTPSession *sess);
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GstFlowReturn rtp_session_produce_rtcp (RTPSession *sess);
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#endif /* __RTP_SESSION_H__ */
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