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600afaaff9
Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads), (async_jitter_queue_pop_intern_unlocked): Fix the case where the buffer underruns and does not block. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): Rename RTCP send pad, like in the session manager. Allow getting an RTCP pad for receiving even if we don't receive RTP. fix handling of send_rtp_src pad. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): When no pt map could be found, fall back to the sinkpad caps. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Fix pad names. * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_create_source), (rtp_session_process_sr), (rtp_session_send_rtp), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Unlock session when performing a callback. Add callbacks for the internal session object. Fix sending of RTP packets. first attempt at adding NTP times in the SR packets. Small debug and doc improvements. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Update stats for SR reports.
263 lines
9.6 KiB
C
263 lines
9.6 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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* 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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* @RTPSessionGetTime: callback for returning the current time
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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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RTPSessionClockRate clock_rate;
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RTPSessionGetTime get_time;
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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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};
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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);
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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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/* 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);
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#endif /* __RTP_SESSION_H__ */
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