gstreamer/gst/rtpmanager/rtpsource.h
Wim Taymans b2aa36cb0d gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
Use lock to protect variable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
Reconstruct GST timestamp from RTP timestamps based on measured clock
skew and sync offset.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_set_tail_changed),
(rtp_jitter_buffer_set_clock_rate),
(rtp_jitter_buffer_get_clock_rate), (calculate_skew),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
* gst/rtpmanager/rtpjitterbuffer.h:
Measure clock skew.
Add callback to be notfied when a new packet was inserted at the tail.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Remove clock skew detection, it's move to the jitterbuffer now.
2009-08-11 02:30:30 +01:00

198 lines
6.5 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;
GstCaps *caps;
gint clock_rate;
gint32 seqnum_base;
gint64 clock_base;
GstClockTime bye_time;
GstClockTime last_activity;
GstClockTime last_rtp_activity;
GstClockTime last_rtptime;
GstClockTime last_ntpnstime;
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_update_caps (RTPSource *src, GstCaps *caps);
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, guint64 ntpnstime);
/* 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, GstClockTime 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);
#endif /* __RTP_SOURCE_H__ */