From 1d75a69ccf4b4ff63037cf5b4ddf9491dad7ca4b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 18 Apr 2007 18:58:53 +0000 Subject: configure.ac: Disable rtpmanager for now because it depends on CVS -base. 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. --- gst/rtpmanager/rtpsession.h | 206 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 206 insertions(+) create mode 100644 gst/rtpmanager/rtpsession.h (limited to 'gst/rtpmanager/rtpsession.h') diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h new file mode 100644 index 00000000..46062c99 --- /dev/null +++ b/gst/rtpmanager/rtpsession.h @@ -0,0 +1,206 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __RTP_SESSION_H__ +#define __RTP_SESSION_H__ + +#include +#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, GstBuffer *buffer, 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); + +/** + * 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); + +/** + * RTPSessionGetTime: + * @sess: an #RTPSession + * @user_data: user data specified when registering + * + * This callback will be called when @sess needs the current time in + * nanoseconds. + * + * Returns: a #GstClockTime with the current time in nanoseconds. + */ +typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data); + +/** + * RTPSessionCallbacks: + * @RTPSessionProcessRTP: callback to process RTP packets + * @RTPSessionSendRTP: callback for sending RTP packets + * @RTPSessionSendRTCP: callback for sending RTCP packets + * @RTPSessionGetTime: callback for returning the current time + * + * 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; + RTPSessionSendRTCP send_rtcp; + RTPSessionClockRate clock_rate; + RTPSessionGetTime get_time; +} RTPSessionCallbacks; + +/** + * RTPSession: + * @lock: lock to protect the session + * @source: the source of this session + * @ssrcs: Hashtable of sources indexed by SSRC + * @cnames: Hashtable of sources indexed by CNAME + * @num_sources: the number of sources + * @activecount: the number of active sources + * @callbacks: callbacks + * @user_data: user data passed in callbacks + * + * The RTP session manager object + */ +struct _RTPSession { + GObject object; + + GMutex *lock; + + guint header_len; + + RTPSource *source; + GHashTable *ssrcs; + GHashTable *cnames; + guint total_sources; + + RTPSessionCallbacks callbacks; + gpointer user_data; + + RTPSessionStats stats; +}; + +/** + * 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; + + /* 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_bye_ssrc) (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_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); + +/* handling sources */ +gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src); +gint rtp_session_get_num_sources (RTPSession *sess); +gint rtp_session_get_num_active_sources (RTPSession *sess); +RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc); +RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname); +RTPSource* rtp_session_create_source (RTPSession *sess); + +/* processing packets from receivers */ +GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer); +GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer); + +/* processing packets for sending */ +GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer); + +/* get interval for next RTCP interval */ +gdouble rtp_session_get_rtcp_interval (RTPSession *sess); +GstFlowReturn rtp_session_produce_rtcp (RTPSession *sess); + +#endif /* __RTP_SESSION_H__ */ -- cgit v1.2.1