mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 05:12:09 +00:00
a468f02d2a
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider): Move reconsideration code to the rtpsession object. Simplify timout handling and add reconsideration. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks), (obtain_source), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_bye), (rtp_session_process_rtcp), (calculate_rtcp_interval), (rtp_session_send_bye), (rtp_session_next_timeout), (session_start_rtcp), (session_report_blocks), (session_cleanup), (session_sdes), (session_bye), (is_rtcp_time), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Handle timeout of inactive sources and senders. Implement BYE scheduling. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (rtp_source_process_sr), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add members to check for timeouts. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter), (rtp_stats_calculate_bye_interval): * gst/rtpmanager/rtpstats.h: Use RFC algorithm for calculating the reporting interval.
185 lines
5.8 KiB
C
185 lines
5.8 KiB
C
/* GStreamer
|
|
* Copyright (C) <2007> Wim Taymans <wim@fluendo.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)
|
|
#define RTP_MAX_DROPOUT 3000
|
|
#define RTP_MAX_MISORDER 100
|
|
|
|
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;
|
|
|
|
/**
|
|
* RTPSource:
|
|
*
|
|
* A source in the #RTPSession
|
|
*/
|
|
struct _RTPSource {
|
|
GObject object;
|
|
|
|
/*< private >*/
|
|
guint32 ssrc;
|
|
|
|
gint probation;
|
|
gboolean validated;
|
|
gboolean is_csrc;
|
|
gboolean is_sender;
|
|
|
|
gchar *cname;
|
|
gchar *name;
|
|
gchar *email;
|
|
gchar *phone;
|
|
gchar *location;
|
|
gchar *tool;
|
|
gchar *note;
|
|
gboolean received_bye;
|
|
gchar *bye_reason;
|
|
|
|
gboolean have_rtp_from;
|
|
GstNetAddress rtp_from;
|
|
gboolean have_rtcp_from;
|
|
GstNetAddress rtcp_from;
|
|
|
|
guint8 payload;
|
|
gint clock_rate;
|
|
|
|
GstClockTime bye_time;
|
|
GstClockTime last_activity;
|
|
GstClockTime last_rtp_activity;
|
|
|
|
GQueue *packets;
|
|
|
|
RTPSourceCallbacks callbacks;
|
|
gpointer user_data;
|
|
|
|
RTPSourceStats stats;
|
|
};
|
|
|
|
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);
|
|
void rtp_source_set_as_csrc (RTPSource *src);
|
|
|
|
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, GstBuffer *buffer);
|
|
|
|
/* RTCP messages */
|
|
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
|
|
void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime,
|
|
guint32 packet_count, guint32 octet_count, GstClockTime time);
|
|
void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost,
|
|
guint32 exthighestseq, guint32 jitter,
|
|
guint32 lsr, guint32 dlsr);
|
|
|
|
gboolean rtp_source_get_last_sr (RTPSource *src, guint64 *ntptime, guint32 *rtptime,
|
|
guint32 *packet_count, guint32 *octet_count, GstClockTime *time);
|
|
gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
|
|
guint32 *exthighestseq, guint32 *jitter,
|
|
guint32 *lsr, guint32 *dlsr);
|
|
|
|
#endif /* __RTP_SOURCE_H__ */
|