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gstreamer/gst/rtpmanager/rtpsource.h
Olivier Crête a6dfe96169 rtpsession: Make it possible to favor new sources in case of SSRC conflict 
Add a "favor-new" property that tells the session to favor new sources when
there is a SSRC conflict. This is useful for SIP calls and other such cases
where a remote loop is extremely unlikely.

Fixes #607615
2010-03-10 11:21:19 +01:00

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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RTP_SOURCE_H__
#define __RTP_SOURCE_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/netbuffer/gstnetbuffer.h>
#include "rtpstats.h"
/* the default number of consecutive RTP packets we need to receive before the
* source is considered valid */
#define RTP_NO_PROBATION 0
#define RTP_DEFAULT_PROBATION 2
#define RTP_SEQ_MOD (1 << 16)
typedef struct _RTPSource RTPSource;
typedef struct _RTPSourceClass RTPSourceClass;
#define RTP_TYPE_SOURCE (rtp_source_get_type())
#define RTP_SOURCE(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource))
#define RTP_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass))
#define RTP_IS_SOURCE(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE))
#define RTP_IS_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE))
#define RTP_SOURCE_CAST(src) ((RTPSource *)(src))
/**
* RTP_SOURCE_IS_ACTIVE:
* @src: an #RTPSource
*
* Check if @src is active. A source is active when it has been validated
* and has not yet received a BYE packet.
*/
#define RTP_SOURCE_IS_ACTIVE(src) (src->validated && !src->received_bye)
/**
* RTP_SOURCE_IS_SENDER:
* @src: an #RTPSource
*
* Check if @src is a sender.
*/
#define RTP_SOURCE_IS_SENDER(src) (src->is_sender)
/**
* RTPSourcePushRTP:
* @src: an #RTPSource
* @buffer: the RTP buffer ready for processing
* @user_data: user data specified when registering
*
* This callback will be called when @src has @buffer ready for further
* processing.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer,
gpointer user_data);
/**
* RTPSourceClockRate:
* @src: an #RTPSource
* @payload: a payload type
* @user_data: user data specified when registering
*
* This callback will be called when @src needs the clock-rate of the
* @payload.
*
* Returns: a clock-rate for @payload.
*/
typedef gint (*RTPSourceClockRate) (RTPSource *src, guint8 payload, gpointer user_data);
/**
* RTPSourceCallbacks:
* @push_rtp: a packet becomes available for handling
* @clock_rate: a clock-rate is requested
* @get_time: the current clock time is requested
*
* Callbacks performed by #RTPSource when actions need to be performed.
*/
typedef struct {
RTPSourcePushRTP push_rtp;
RTPSourceClockRate clock_rate;
} RTPSourceCallbacks;
/**
* RTPConflictingAddress:
* @address: #GstNetAddress which conflicted
* @last_conflict_time: time when the last conflict was seen
*
* This structure is used to account for addresses that have conflicted to find
* loops.
*/
typedef struct {
GstNetAddress address;
GstClockTime time;
} RTPConflictingAddress;
/**
* RTPSource:
*
* A source in the #RTPSession
*
* @conflicting_addresses: GList of conflicting addresses
*/
struct _RTPSource {
GObject object;
/*< private >*/
guint32 ssrc;
gint probation;
gboolean validated;
gboolean internal;
gboolean is_csrc;
gboolean is_sender;
GstStructure *sdes;
gboolean received_bye;
gchar *bye_reason;
gboolean have_rtp_from;
GstNetAddress rtp_from;
gboolean have_rtcp_from;
GstNetAddress rtcp_from;
gint payload;
GstCaps *caps;
gint clock_rate;
gint32 seqnum_base;
GstClockTime bye_time;
GstClockTime last_activity;
GstClockTime last_rtp_activity;
GstClockTime last_rtime;
GstClockTime last_rtptime;
/* for bitrate estimation */
guint64 bitrate;
GstClockTime prev_rtime;
guint64 bytes_sent;
guint64 bytes_received;
GQueue *packets;
RTPSourceCallbacks callbacks;
gpointer user_data;
RTPSourceStats stats;
GList *conflicting_addresses;
};
struct _RTPSourceClass {
GObjectClass parent_class;
};
GType rtp_source_get_type (void);
/* managing lifetime of sources */
RTPSource* rtp_source_new (guint32 ssrc);
void rtp_source_set_callbacks (RTPSource *src, RTPSourceCallbacks *cb, gpointer data);
/* properties */
guint32 rtp_source_get_ssrc (RTPSource *src);
void rtp_source_set_as_csrc (RTPSource *src);
gboolean rtp_source_is_as_csrc (RTPSource *src);
gboolean rtp_source_is_active (RTPSource *src);
gboolean rtp_source_is_validated (RTPSource *src);
gboolean rtp_source_is_sender (RTPSource *src);
gboolean rtp_source_received_bye (RTPSource *src);
gchar * rtp_source_get_bye_reason (RTPSource *src);
void rtp_source_update_caps (RTPSource *src, GstCaps *caps);
/* SDES info */
gboolean rtp_source_set_sdes_string (RTPSource *src, GstRTCPSDESType type,
const gchar *data);
gchar* rtp_source_get_sdes_string (RTPSource *src, GstRTCPSDESType type);
const GstStructure *
rtp_source_get_sdes_struct (RTPSource * src);
gboolean rtp_source_set_sdes_struct (RTPSource * src, GstStructure *sdes);
/* handling network address */
void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address);
void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address);
/* handling RTP */
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
GstFlowReturn rtp_source_send_rtp (RTPSource *src, gpointer data, gboolean is_list,
GstClockTime running_time);
/* RTCP messages */
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
void rtp_source_process_sr (RTPSource *src, GstClockTime time, guint64 ntptime,
guint32 rtptime, guint32 packet_count, guint32 octet_count);
void rtp_source_process_rb (RTPSource *src, GstClockTime time, guint8 fractionlost,
gint32 packetslost, guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
gboolean rtp_source_get_new_sr (RTPSource *src, guint64 ntpnstime, GstClockTime running_time,
guint64 *ntptime, guint32 *rtptime, guint32 *packet_count,
guint32 *octet_count);
gboolean rtp_source_get_new_rb (RTPSource *src, GstClockTime time, guint8 *fractionlost,
gint32 *packetslost, guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
gboolean rtp_source_get_last_sr (RTPSource *src, GstClockTime *time, guint64 *ntptime,
guint32 *rtptime, guint32 *packet_count,
guint32 *octet_count);
gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr, guint32 *round_trip);
void rtp_source_reset (RTPSource * src);
gboolean rtp_source_find_conflicting_address (RTPSource * src,
GstNetAddress *address,
GstClockTime time);
void rtp_source_add_conflicting_address (RTPSource * src,
GstNetAddress *address,
GstClockTime time);
void rtp_source_timeout (RTPSource * src,
GstClockTime current_time,
GstClockTime collision_timeout);
#endif /* __RTP_SOURCE_H__ */
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