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gstreamer/gst/rtpmanager/rtpsource.h
Wim Taymans e779adca69 gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first 
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
2009-08-11 02:30:34 +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)
#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;
guint8 *sdes[9];
guint sdes_len[9];
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;
guint64 clock_base_time;
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_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 (RTPSource *src, GstRTCPSDESType type,
const guint8 *data, guint len);
gboolean rtp_source_set_sdes_string (RTPSource *src, GstRTCPSDESType type,
const gchar *data);
gboolean rtp_source_get_sdes (RTPSource *src, GstRTCPSDESType type,
guint8 **data, guint *len);
gchar* rtp_source_get_sdes_string (RTPSource *src, GstRTCPSDESType type);
/* 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, 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);
void rtp_source_reset (RTPSource * src);
#endif /* __RTP_SOURCE_H__ */
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