gstreamer/gst-libs/gst/rtp/gstrtcpbuffer.h
Wim Taymans 6f93db5ab5 Fix parsing of RB blocks.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix parsing of RB blocks.
Fix docs.
Added helper functions to convert to/from UNIX and NTP time.
API: gst_rtcp_ntp_to_unix()
API: gst_rtcp_unix_to_ntp()
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_get_header_len),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix some more docs.
Implement handling of packets with extensions.
Fix padding check in _validate().
Added function to get extension data.
API: gst_rtp_buffer_get_header_len()
API: gst_rtp_buffer_get_extension_data()
2007-09-03 19:31:11 +00:00

242 lines
9 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* gstrtcpbuffer.h: various helper functions to manipulate buffers
* with RTCP payload.
*
* 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 __GST_RTCPBUFFER_H__
#define __GST_RTCPBUFFER_H__
#include <gst/gst.h>
G_BEGIN_DECLS
/**
* GST_RTCP_VERSION:
*
* The supported RTCP version 2.
*/
#define GST_RTCP_VERSION 2
/**
* GstRTCPType:
* @GST_RTCP_TYPE_INVALID: Invalid type
* @GST_RTCP_TYPE_SR: Sender report
* @GST_RTCP_TYPE_RR: Receiver report
* @GST_RTCP_TYPE_SDES: Source description
* @GST_RTCP_TYPE_BYE: Goodbye
* @GST_RTCP_TYPE_APP: Application defined
*
* Different RTCP packet types.
*/
typedef enum
{
GST_RTCP_TYPE_INVALID = 0,
GST_RTCP_TYPE_SR = 200,
GST_RTCP_TYPE_RR = 201,
GST_RTCP_TYPE_SDES = 202,
GST_RTCP_TYPE_BYE = 203,
GST_RTCP_TYPE_APP = 204
} GstRTCPType;
/**
* GstRTCPSDESType:
* @GST_RTCP_SDES_INVALID: Invalid SDES entry
* @GST_RTCP_SDES_END: End of SDES list
* @GST_RTCP_SDES_CNAME: Canonical name
* @GST_RTCP_SDES_NAME: User name
* @GST_RTCP_SDES_EMAIL: User's electronic mail address
* @GST_RTCP_SDES_PHONE: User's phone number
* @GST_RTCP_SDES_LOC: Geographic user location
* @GST_RTCP_SDES_TOOL: Name of application or tool
* @GST_RTCP_SDES_NOTE: Notice about the source
* @GST_RTCP_SDES_PRIV: Private extensions
*
* Different types of SDES content.
*/
typedef enum
{
GST_RTCP_SDES_INVALID = -1,
GST_RTCP_SDES_END = 0,
GST_RTCP_SDES_CNAME = 1,
GST_RTCP_SDES_NAME = 2,
GST_RTCP_SDES_EMAIL = 3,
GST_RTCP_SDES_PHONE = 4,
GST_RTCP_SDES_LOC = 5,
GST_RTCP_SDES_TOOL = 6,
GST_RTCP_SDES_NOTE = 7,
GST_RTCP_SDES_PRIV = 8
} GstRTCPSDESType;
/**
* GST_RTCP_MAX_SDES:
*
* The maximum text length for an SDES item.
*/
#define GST_RTCP_MAX_SDES 255
/**
* GST_RTCP_MAX_RB_COUNT:
*
* The maximum amount of Receiver report blocks in RR and SR messages.
*/
#define GST_RTCP_MAX_RB_COUNT 31
/**
* GST_RTCP_MAX_SDES_ITEM_COUNT:
*
* The maximum amount of SDES items.
*/
#define GST_RTCP_MAX_SDES_ITEM_COUNT 31
/**
* GST_RTCP_MAX_BYE_SSRC_COUNT:
*
* The maximum amount of SSRCs in a BYE packet.
*/
#define GST_RTCP_MAX_BYE_SSRC_COUNT 31
/**
* GST_RTCP_VALID_MASK:
*
* Mask for version, padding bit and packet type pair
*/
#define GST_RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe)
/**
* GST_RTCP_VALID_VALUE:
*
* Valid value for the first two bytes of an RTCP packet after applying
* #GST_RTCP_VALID_MASK to them.
*/
#define GST_RTCP_VALID_VALUE ((GST_RTCP_VERSION << 14) | GST_RTCP_TYPE_SR)
typedef struct _GstRTCPPacket GstRTCPPacket;
/**
* GstRTCPPacket:
* @buffer: pointer to RTCP buffer
* @offset: offset of packet in buffer data
*
* Data structure that points to a packet at @offset in @buffer.
* The size of the structure is made public to allow stack allocations.
*/
struct _GstRTCPPacket
{
GstBuffer *buffer;
guint offset;
/*< private >*/
gboolean padding; /* padding field of current packet */
guint8 count; /* count field of current packet */
GstRTCPType type; /* type of current packet */
guint16 length; /* length of current packet in 32-bits words */
guint item_offset; /* current item offset for navigating SDES */
guint item_count; /* current item count */
guint entry_offset; /* current entry offset for navigating SDES items */
};
/* creating buffers */
GstBuffer* gst_rtcp_buffer_new_take_data (gpointer data, guint len);
GstBuffer* gst_rtcp_buffer_new_copy_data (gpointer data, guint len);
gboolean gst_rtcp_buffer_validate_data (guint8 *data, guint len);
gboolean gst_rtcp_buffer_validate (GstBuffer *buffer);
GstBuffer* gst_rtcp_buffer_new (guint mtu);
void gst_rtcp_buffer_end (GstBuffer *buffer);
/* adding/retrieving packets */
guint gst_rtcp_buffer_get_packet_count (GstBuffer *buffer);
gboolean gst_rtcp_buffer_get_first_packet (GstBuffer *buffer, GstRTCPPacket *packet);
gboolean gst_rtcp_packet_move_to_next (GstRTCPPacket *packet);
gboolean gst_rtcp_buffer_add_packet (GstBuffer *buffer, GstRTCPType type,
GstRTCPPacket *packet);
void gst_rtcp_packet_remove (GstRTCPPacket *packet);
/* working with packets */
gboolean gst_rtcp_packet_get_padding (GstRTCPPacket *packet);
guint8 gst_rtcp_packet_get_count (GstRTCPPacket *packet);
GstRTCPType gst_rtcp_packet_get_type (GstRTCPPacket *packet);
guint16 gst_rtcp_packet_get_length (GstRTCPPacket *packet);
/* sender reports */
void gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket *packet, guint32 *ssrc,
guint64 *ntptime, guint32 *rtptime,
guint32 *packet_count, guint32 *octet_count);
void gst_rtcp_packet_sr_set_sender_info (GstRTCPPacket *packet, guint32 ssrc,
guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count);
/* receiver reports */
guint32 gst_rtcp_packet_rr_get_ssrc (GstRTCPPacket *packet);
void gst_rtcp_packet_rr_set_ssrc (GstRTCPPacket *packet, guint32 ssrc);
/* report blocks for SR and RR */
guint gst_rtcp_packet_get_rb_count (GstRTCPPacket *packet);
void gst_rtcp_packet_get_rb (GstRTCPPacket *packet, guint nth, guint32 *ssrc,
guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
gboolean gst_rtcp_packet_add_rb (GstRTCPPacket *packet, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
void gst_rtcp_packet_set_rb (GstRTCPPacket *packet, guint nth, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
/* source description packet */
guint gst_rtcp_packet_sdes_get_item_count (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_first_item (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_next_item (GstRTCPPacket *packet);
guint32 gst_rtcp_packet_sdes_get_ssrc (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_first_entry (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_next_entry (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_get_entry (GstRTCPPacket *packet,
GstRTCPSDESType *type, guint8 *len,
guint8 **data);
gboolean gst_rtcp_packet_sdes_copy_entry (GstRTCPPacket *packet,
GstRTCPSDESType *type, guint8 *len,
guint8 **data);
gboolean gst_rtcp_packet_sdes_add_item (GstRTCPPacket *packet, guint32 ssrc);
gboolean gst_rtcp_packet_sdes_add_entry (GstRTCPPacket *packet, GstRTCPSDESType type,
guint8 len, const guint8 *data);
/* bye packet */
guint gst_rtcp_packet_bye_get_ssrc_count (GstRTCPPacket *packet);
guint32 gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket *packet, guint nth);
gboolean gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket *packet, guint32 ssrc);
gboolean gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket *packet, guint32 *ssrc, guint len);
guint8 gst_rtcp_packet_bye_get_reason_len (GstRTCPPacket *packet);
gchar* gst_rtcp_packet_bye_get_reason (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_bye_set_reason (GstRTCPPacket *packet, const gchar *reason);
/* helper functions */
guint64 gst_rtcp_ntp_to_unix (guint64 ntptime);
guint64 gst_rtcp_unix_to_ntp (guint64 unixtime);
G_END_DECLS
#endif /* __GST_RTCPBUFFER_H__ */