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e7b6212c51
Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: Updated example pipelines in docs. Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync. Set the default latency correctly. Add some more points where we can get caps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Add ts-offset property to control timestamping. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Various cleanups. Feed rtpsession manager with NTP time based on pipeline clock when handling RTP packets and RTCP timeouts. Perform all RTCP with the system clock. Set caps on RTCP outgoing buffers. * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc), (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain): * gst/rtpmanager/gstrtpssrcdemux.h: Also demux RTCP messages. * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_rb), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_send_bye), (session_start_rtcp), (session_report_blocks), (session_cleanup), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Remove the get_time callback, the GStreamer part will feed us with enough timing information. Split sync timing and RTCP timing information. Factor out common RB handling for SR and RR. Send out SR RTCP packets for lip-sync. Move SR and RR packet info generation to the source. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_new_sr), (rtp_source_get_new_rb), (rtp_source_get_last_sr): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Use caps on incomming buffers to get timing information when they are there. Calculate clock scew of the receiver compared to the sender and adjust the rtp timestamps. Calculate the round trip in sources. Do SR and RR calculations in the source.
268 lines
9.8 KiB
C
268 lines
9.8 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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* RTPSessionSyncRTCP:
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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 and SR @buffer ready for doing
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* synchronisation between streams.
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*
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* Returns: a #GstFlowReturn.
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*/
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typedef GstFlowReturn (*RTPSessionSyncRTCP) (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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* RTPSessionReconsider:
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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 to cancel the current timeout.
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* The currently running timeout should be canceled and a new reporting interval
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* should be requested from @sess.
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*/
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typedef void (*RTPSessionReconsider) (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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* @RTPSessionSyncRTCP: callback for handling SR packets
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* @RTPSessionReconsider: callback for reconsidering the timeout
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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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RTPSessionSyncRTCP sync_rtcp;
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RTPSessionClockRate clock_rate;
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RTPSessionReconsider reconsider;
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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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guint mtu;
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RTPSource *source;
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/* info for creating reports */
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gchar *cname;
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gchar *name;
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gchar *email;
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gchar *phone;
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gchar *location;
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gchar *tool;
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gchar *note;
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/* for sender/receiver counting */
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guint32 key;
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guint32 mask_idx;
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guint32 mask;
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GHashTable *ssrcs[32];
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GHashTable *cnames;
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guint total_sources;
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GstClockTime next_rtcp_check_time;
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GstClockTime last_rtcp_send_time;
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gboolean first_rtcp;
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GstBuffer *bye_packet;
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gchar *bye_reason;
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gboolean sent_bye;
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RTPSessionCallbacks callbacks;
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gpointer user_data;
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RTPSessionStats stats;
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/* for mapping clock time to NTP time */
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GstClockTime base_time;
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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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void (*on_bye_timeout) (RTPSession *sess, RTPSource *source);
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void (*on_timeout) (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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void rtp_session_set_cname (RTPSession *sess, const gchar *cname);
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gchar* rtp_session_get_cname (RTPSession *sess);
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void rtp_session_set_name (RTPSession *sess, const gchar *name);
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gchar* rtp_session_get_name (RTPSession *sess);
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void rtp_session_set_email (RTPSession *sess, const gchar *email);
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gchar* rtp_session_get_email (RTPSession *sess);
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void rtp_session_set_phone (RTPSession *sess, const gchar *phone);
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gchar* rtp_session_get_phone (RTPSession *sess);
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void rtp_session_set_location (RTPSession *sess, const gchar *location);
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gchar* rtp_session_get_location (RTPSession *sess);
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void rtp_session_set_tool (RTPSession *sess, const gchar *tool);
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gchar* rtp_session_get_tool (RTPSession *sess);
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void rtp_session_set_note (RTPSession *sess, const gchar *note);
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gchar* rtp_session_get_note (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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guint rtp_session_get_num_sources (RTPSession *sess);
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guint 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, guint64 ntpnstime);
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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, guint64 ntptime);
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/* stopping the session */
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GstFlowReturn rtp_session_send_bye (RTPSession *sess, const gchar *reason);
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/* get interval for next RTCP interval */
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GstClockTime rtp_session_next_timeout (RTPSession *sess, GstClockTime time);
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GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTime time, guint64 ntpnstime);
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#endif /* __RTP_SESSION_H__ */
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